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Old 26th July 2014, 01:06   #7601  |  Link
VideoFanatic
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MeGUI causes Delay in Audio

Log. Forgot to add -tff switch so video was 1080p instead of 1080i but even when I do it correctly whatever resolution I encode to I get a delay.

Anything I encode with MeGUI 2507 has an audio delay. I'm converting 1080i 25fps videos to the same resolution and to 720p 50fps with MeGUI. The videos were cut with VideoRedo and from experience I know that my video sources don't have a delay in them. However converting the 1080i videos to the same resolution or to 720p with MeGUI causes a 65 second delay in the audio, MeGUI is causing that.

I know MeGUI is causing that because I could demux and remux my source with DGIndexNV/TSmuxer and there would be no delay.

I ticked the option in MeGUI to use DGDecodeNV.
I chose the option to never re-encode the AC3 128 Kbps audio that's in my sources.

It doesn't matter what container I encode to with MeGUI, I always get a delay.

I can convert a 45 minute episode fine, there's still a delay but it doesn't cause the audio to go out of sync. Yet with a 2 hour 15 minute video or more, I always get a 65 second delay.

I use these advanced settings to make the video Bluray compliant:

Code:
--crf 17 --preset veryfast --bluray-compat --vbv-maxrate 15000 --vbv-bufsize 15000 --profile high --level 4.0 --slices 1 --sar 1:1
I chose High Profile in the GUI. I also tried GOP Calculation: FPS based and it uses 500 KeyInt. I tried Fixed and 100 KeyInt so it does actually use 100 KeyInt but it made no difference, I still get a delay.

Not to complicate things further, for testing purposes, I'm using this basic Aviysnth script for converting 1080i files to the same resolution:

Code:
setmtmode(5,9)
<input>
setmtmode(2)
Tweak(Hue=0, Sat=1.0, Bright=0, Cont=1.0, Coring=False)
How can I prevent my MeGUI encodings from getting a delay?

Last edited by VideoFanatic; 26th July 2014 at 01:24.
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Old 26th July 2014, 01:22   #7602  |  Link
nekrovski
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Regarding my last post I fixed the problem with installing matroska splitter. It probably got messed up (read uninstalled) when I was going through a guide for codecs/mpc-hc/madvr etc
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Old 26th July 2014, 19:02   #7603  |  Link
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holygamer,
From your log file:

---[Information] [25/07/2014 21:36:42] File: E:\1 = Working Videos\WWE 2014\Raw from July\1v5ymxyt.v2w\Raw Mar 24 2014 - [0] English Delay 65571ms.ac3

As far as I know, MeGUI uses the audio delay reported by MediaInfo. In your case it's 65571ms. I don't know why MediaInfo is reporting such an odd delay, but given you're first cutting the videos using VideoRedo, I'd guess that's where something odd is happening. MeGUI writes the reported delay to the extracted audio's file name and it's automatically applied when muxing. Unless someone has a better idea, maybe the best you can do is mux the encoded video/audio manually without applying the delay.

Can you use MeGUI to do the cutting? If you create a script you can use the AVS Cutter under the Tools menu to specify multiple start and end points for encoding, effectively cutting/editing as you encode. The AVS Cutter can save a "cuts file" which can be loaded into the audio section in order to re-encode it to match, or the audio cutter under the Tools menu can cut/rejoin extracted audio without re-encoding it.
You appear to be using the OneClick encoder. I don't use it myself but I assume the above method would be best done "manually". ie use the File/Open menu to open a file, index it and extract the audio, create a script and then add the cuts etc.....

Where did you get the x264 settings you use for Bluray compatibility?
If you open the x264 encoder configuration, load the defaults, then select Bluray as the target playback device from the dropdown list, you get this:

Quote:
program --level 4.1 --bluray-compat --open-gop --slices 4 --vbv-bufsize 30000 --vbv-maxrate 40000 --output "output" "input"
You can of course change the x264 speed preset. Open Gop isn't compulsory but it's probably a good idea for Bluray (due to the short key interval). MeGUI will automatically make a few adjustments behind the scenes to settings such as -keyint. Those settings are based on the frame rate so they don't appear in the initial x264 commandline, but they'll be specified in the log file.
You shouldn't need --sar 1:1. As long as you're not using anamorphic encoding and you're just resizing, the sar will always be 1:1.
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Old 26th July 2014, 19:24   #7604  |  Link
kalehrl
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When an audio track is demuxed from a mkv, audio delay is written in the file name and is applied when muxing.
However, when you take an audio track with a delay and transcode it to a different format, the audio delay is not added to the file name of the new file.
This seems to be a small bug.
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Old 26th July 2014, 19:30   #7605  |  Link
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Thanks. I never even thought to look in MediaInfo for a delay. You're correct there's a delay mentioned there. There's no delay mentioned in MediaInfo on a VideoRedo edited file. It's only when I remux the video and convert the audio from AAC to AC3 with FFMPEG that the muxed file gets a delay. Muxing from TS to TS gets a -5ms delay. Muxing from MKV to MKV gets a 1 minute 5 second delay!

I didn't get a delay in 45 minute files because I didn't use FFMPEG on those files.

Here's my FFMPEG script:
Code:
"X:\Portable Installations\ffmpeg-2014 May 14-git-72dcd48-win64-static\bin\ffmpeg.exe" -i %1 -c copy -absf aac_adtstoasc -vn gain.m4a
"X:\Portable Installations\AAC Gain 1.9\aacgain" -r -k -m 0 gain.m4a
"X:\Portable Installations\ffmpeg-2014 May 14-git-72dcd48-win64-static\bin\ffmpeg.exe" -i gain.m4a -acodec copy "%~d1%~p1%~n1.aac"
"X:\Portable Installations\ffmpeg-2014 May 14-git-72dcd48-win64-static\bin\ffmpeg.exe" -i %1 -i gain.m4a -map 0:v -map 1 -c:v copy -c:a ac3 -ab 2 -ab 128k "%~d1%~p1%~n1.AC3.mkv"
del gain.m4a
"X:\Daves Folder\Sounds\VideoRedo Completed Sound Short.WAV"
pause
Basically I use the Elgato Game Capture HD to capture Satellite TV. Annoyingly it uses AAC instead of AC3 which isn't Bluray compliant. So I put the above batch file in the "Send To" directory in Windows, I right-click a video, select Send TO then select the batch file and it does the following: Muxes MKV to MKV and re-encodes AAC to AC3. It also demuxes the AAC in the MKV into an AAC file and normalizes both the AAC and AC3!

Is there a script I can use to prevent FFMPEG adding a delay to the muxed file it makes? Or is there a way to tell MeGUI to ignore the audio delay?

With regards to my Bluray switches, I just did what was required to make a compliant disc as mentioned on this thread and the Doom9 guide: http://forum.videohelp.com/threads/3...Disc?p=2335130

Last edited by VideoFanatic; 26th July 2014 at 19:39.
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Old 27th July 2014, 03:15   #7606  |  Link
hello_hello
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Quote:
Originally Posted by kalehrl View Post
When an audio track is demuxed from a mkv, audio delay is written in the file name and is applied when muxing.
However, when you take an audio track with a delay and transcode it to a different format, the audio delay is not added to the file name of the new file.
This seems to be a small bug.
I ran a test encode and it seems MeGUI compensates for any audio delay when encoding. I'm not sure when it started doing that (maybe it always has), or even if it's a good idea (it probably is most of the time) but when I added an audio file named "test - [0] English Delay 18ms" to MeGUI's audio section, it automatically named the output "test - [0] English Delay 0ms".

I checked the output file and it was indeed 18ms longer than the original. A look through the log file revealed MeGUI added the following to the Avisynth script:
DelayAudio(18.0/1000.0)

I rarely use MeGUI for the audio encoding myself so I wasn't aware it does that. It's handy to know though.

Last edited by hello_hello; 27th July 2014 at 03:52.
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Old 27th July 2014, 03:49   #7607  |  Link
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Quote:
Originally Posted by holygamer View Post
Is there a script I can use to prevent FFMPEG adding a delay to the muxed file it makes? Or is there a way to tell MeGUI to ignore the audio delay?
I don't use ffmpeg so I can't help there. Is there another thread where you could ask that question (ie an ffmpeg thread)? It seems like it's either an ffmpeg bug or for some reason MediaInfo is getting it wrong.
I don't think there's any way to tell MeGUI to ignore the delay. Doing things the "manual" way you could simply change the delay written to the audio file's name after it's extracted. Or do the muxing manually. MKVMergeGUI (or MeGUI's muxers) will automatically apply the audio delay written to the file name, but you can manually change it when muxing.
Using the Oneclick encoder, I'm not sure.

Quote:
Originally Posted by holygamer View Post
With regards to my Bluray switches, I just did what was required to make a compliant disc as mentioned on this thread and the Doom9 guide: http://forum.videohelp.com/threads/3...Disc?p=2335130
Sorry, I'd forgotten you could use --slices 1 if you use lower vbv values etc, although I'm not sure if --slices 4 with the higher VBV values wouldn't be better.
How do you normally set --keyint? You may already be doing it this way (I'm not sure from your earlier post) but if you select "fps based" in MeGUI's encoder configuration, maximum GOP 50 and minimum GOP 25, you should end up with the correct --keyint setting based on the frame rate (2 second GOP, matching the other Bluray compatible settings you use).

ie if the frame rate is 24fps,
50/25*24 = keyint 48

Last edited by hello_hello; 27th July 2014 at 04:01.
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Old 27th July 2014, 10:08   #7608  |  Link
Sharc
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Quote:
Originally Posted by hello_hello View Post
holygamer,
.... Where did you get the x264 settings you use for Bluray compatibility?.....
Note that his VBV settings are perfectly right if he wants to burn to SL disks (BD5, BD9). Otherwise, a reference for blu-ray compliant settings can be found here.
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Old 27th July 2014, 10:30   #7609  |  Link
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Ah yes..... it's coming back to me. I don't encode for Bluray compliance myself. I forgot the lower VBV settings are for burning to DVD discs.
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Old 29th July 2014, 21:38   #7610  |  Link
leon
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Hi
how can I fix this problem:



Avisynth script:
Code:
LoadPlugin("...")
FFAudioSource("...")
# detected channels: 2 channels
# detected channel positions: 1/0/0
Mediainfo:
Code:
Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 1h 1mn
Bit rate mode                            : Constant
Bit rate                                 : 64.0 Kbps
Channel(s)                               : 2 channels
Channel(s)_Original                      : 1 channel
Channel positions                        : Front: C
Sampling rate                            : 44.1 KHz
Compression mode                         : Lossy
Stream size                              : 28.4 MiB (10%)
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Old 30th July 2014, 10:58   #7611  |  Link
tebasuna51
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@leon

I can't reproduce your problem:

Quote:
--[Information] [30/07/2014 10:42:44] Trying to open the file with Import()
--[Information] [30/07/2014 10:42:47] Successfully opened the file with Import()
--[Information] [30/07/2014 10:42:54] ignoring downmix as there is only 2 channel
--[Information] [30/07/2014 10:42:54] Avisynth script
---[NoImage] Import("D:\test.avs")
---[NoImage] # detected channels: 2 channels
---[NoImage] # detected channel positions:
---[NoImage] SSRC(96000)
---[NoImage] Normalize()
...
The avs:

Quote:
LoadPlugin("D:\Programa\megui\tools\ffms\ffms2.dll")
FFAudioSource("D:\test.mp4")
Mediainfo:

Quote:
Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : HE-AACv2 / HE-AAC / LC
Codec ID : 40
Duration : 20s 153ms
Bit rate mode : Variable
Bit rate : 64.0 Kbps
Maximum bit rate : 68.4 Kbps
Channel(s) : 2 channels / 1 channel / 1 channel
Channel positions : Front: L R / Front: C / Front: C
Sampling rate : 44.1 KHz / 44.1 KHz / 22.05 KHz
Maybe your audio have some problem to be recognized properly like SBR-PS (high frequencies in SBR part, second channel in PS part).
Please put a sample.

BTW I can't understand your upsample to 96 KHz (SSRC)
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Old 30th July 2014, 20:14   #7612  |  Link
hello_hello
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It seems to happen (sometimes) if the input audio is mono. I tried converting a mono flac file and that was fine, but a mono m4a file containing AAC produced a similar warning. The output file was stereo (dual mono). It seems under some circumstances, mono audio is decoded to stereo, hence the warning. At least that's how it appears. I'm not sure it's anything to worry about unless you particularly want a single channel output.

leon's MediaInfo report..... I don't know whether that'd be the input or the output file or why it reports Channel(s) and Channel(s)_Original. He wasn't terribly specific.

Quote:
--[Information] [31/07/14 3:57:58 AM] Trying to open the file with NicAudio
--[Information] [31/07/14 3:57:58 AM] Failed opening the file with NicAudio()
--[Information] [31/07/14 3:57:58 AM] Trying to open the file with BassAudioSource()
--[Information] [31/07/14 3:58:01 AM] Successfully opened the file with BassAudioSource()
--[Warning] [31/07/14 3:58:03 AM] channel count mismatch! The input file is reporting 1 channels and the AviSynth script is reporting 2 channels
--[Information] [31/07/14 3:58:03 AM] Avisynth script
---[NoImage] ClearAutoloadDirs()
---[NoImage] AddAutoloadDir("C:\Program Files\MeGUI\tools\avs\plugins")
---[NoImage] LoadPlugin("C:\Program Files\MeGUI\tools\avisynth_plugin\BassAudio.dll")
---[NoImage] BassAudioSource("D:\test.m4a")
---[NoImage] # detected channels: 1 channel
---[NoImage] # detected channel positions: 1/0/0
---[NoImage] return last
--[Information] [31/07/14 3:58:03 AM] Commandline used: -ignorelength -q 0.5 -if - -of "{0}"
--[Information] [31/07/14 3:58:03 AM] Avisynth script environment opened
--[Information] [31/07/14 3:58:03 AM] Script loaded
--[Information] [31/07/14 3:58:03 AM] Output Decoder
---[Information] [31/07/14 3:58:03 AM] Channels: 2
---[Information] [31/07/14 3:58:03 AM] Bits per sample: 32
---[Information] [31/07/14 3:58:03 AM] Sample rate: 48000
--[Information] [31/07/14 3:58:03 AM] Job commandline: C:\Program Files\MeGUI\tools\eac3to\neroAacEnc.exe -ignorelength -q 0.5 -if - -of "D:\test_new.m4a"
--[Information] [31/07/14 3:58:03 AM] Process started
--[Information] [31/07/14 3:58:03 AM] Standard output stream
--[Information] [31/07/14 3:58:03 AM] Standard error stream
--[Information] [31/07/14 3:58:15 AM] MediaInfo
---[Information] [31/07/14 3:58:15 AM] File: D:\test_new.m4a
---[Information] General
----[Information] Format: MPEG-4
----[Information] FormatString: MPEG-4
----[Information] FileSize: 3241936
----[Information] PlayTime: 00:03:49.909
---[Information] Audio
----[Information] ID: 1
----[Information] StreamOrder: 0
----[Information] Format: AAC
----[Information] FormatProfile: LC
----[Information] FormatSettingsSBR: No (Explicit)
----[Information] FormatSettingsPS:
----[Information] SamplingRate: 48000
----[Information] SamplingRateString: 48.0 KHz
----[Information] Channels: 2
----[Information] ChannelsString: 2 channels
----[Information] ChannelPositionsString2: 2/0/0
----[Information] BitRateMode: VBR
----[Information] Delay:
----[Information] Title:
----[Information] Language:
----[Information] LanguageString:
----[Information] Default:
----[Information] DefaultString:
----[Information] Forced:
----[Information] ForcedString:
--[Information] [31/07/14 3:58:15 AM] Postprocessing
--[Information] [31/07/14 3:58:15 AM] Job completed
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Old 31st July 2014, 00:09   #7613  |  Link
leon
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@tebasuna51
Hi thank you for your help.
I think the original audio was mono and it's been upmixed but wrong.
in MeGUI log window detected channels is one but in the script it's 2 this applies to channel positions to.
if MeGUI loads what is in scripts so why they aren't match.
I finally converted the audio with "Xilisoft Video Converter" to WAVE 2 channels and then encoded it with QAAC.
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Old 31st July 2014, 00:15   #7614  |  Link
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@hello_hello

Hi
it's the input (original) audio,the output reports channel as mono.
your detected channels and channel positions are correct but look at mine how could one detected channel be 2/0/0.
I use downmix to Dolby pro logic II maybe that's the problem.
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Old 31st July 2014, 03:03   #7615  |  Link
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Maybe I'm on the wrong track, but the info as to which channel is which is obviously saved as part of the audio stream (left, right, centre etc) so maybe it's possible for a mono channel to be configured as left and right rather than as centre only. That'd explain:

Channel(s) : 2 channels
Channel(s)_Original : 1 channel

in the MediaInfo report for the original file. If the output is reported as mono, maybe that's "correct" in respect to having the same number of channels?

The MeGUI log file.... before it tries to commence encoding, shows a MediaInfo report for the file. In my case it always includes this for a mono AAC input file (under MediaInfo/Audio):

[Information] Channels: 1
[Information] ChannelsString: 1 channel
[Information] ChannelPositionsString2: 1/0/0

For other formats such as MP3 the ChannelPositionsString2 field is empty. Does your input file look any different?

I'll confess I'm guessing, but there does seem to be some inconsistencies when I tested some encodes of a mono file. If the source was mka, MP3 or an ffaudiosource script, the output was mono with no warnings. The same AAC audio in an m4a and the log file included the channel mismatch warning, the decoder output as 2 channels and the output file was stereo. Which seems odd, as the MediaInfo report was the same as above each time. The difference there seems to be the decoder used. It was NicAudio for the mka, BassAudio for the m4a.

Anyway, sorry..... I guess the channel mismatch I'm seeing in the log file is caused by a different circumstance than yours, but maybe I've helped someone work out why it's happening....?
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Old 31st July 2014, 07:47   #7616  |  Link
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I recently had a discussion about the M4A header created by QAAC which may indeed be slightly confusing. According to QAAC's author, nu774, there is an obsolete flag which only reports if the audio is mono or stereo (similar to MPEG1 audio), but most software is supposed to ignore it, because the AAC audio stream "kernel" has own flags which are more modern and detailed, as useful as e.g. WAVEFORMATEXTENSIBLE or Dolby Digital AC3 channel flags, supporting multi-channel audio configurations. Here in the HydrogenAudio forum:

Quote:
Originally Posted by nu774
mp4a.channelcount is intentionally set to either 1 or 2, since in the former spec (ISO 14496-12) only 1 or 2 was allowed. However, it seems that no such restriction present in the new spec (4th ed @2012-7-15). Therefore, maybe I should fix qaac to write actual number of channels.

On the other hand, I can say for sure that nothing in AudioSampleEntry (mp4a) matters and cannot be taken seriously.
For example, samplesize field (typically it is 16) is non-sense for AAC.
samplerate field is 16.16 fixed-point, so not enough to represent high samplerate such as 96kHz.
I don't understand why MediaInfo looks at it at all.
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Old 31st July 2014, 08:51   #7617  |  Link
leon
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Hi
MeGUI is good with 5.1 channel and it can detect channels ans channel positions.
with 2 channels most of the time can't detect positions.
and it's not good with mono but when I convert my audio to WAVE not matter 1,2 or 5.1 channel it works good.
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Old 31st July 2014, 23:53   #7618  |  Link
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Quote:
Originally Posted by LigH View Post
I recently had a discussion about the M4A header created by QAAC which may indeed be slightly confusing. According to QAAC's author, nu774, there is an obsolete flag which only reports if the audio is mono or stereo (similar to MPEG1 audio), but most software is supposed to ignore it, because the AAC audio stream "kernel" has own flags which are more modern and detailed, as useful as e.g. WAVEFORMATEXTENSIBLE or Dolby Digital AC3 channel flags, supporting multi-channel audio configurations.
I guess that explains the "oddness" displayed by MediaInfo I noticed recently. I generally use NeroAAC but I was running some comparison encodes (using foobar2000) and noticed MediaInfo reported a 5.1ch AAC stream encoded with QAAC as:

Channel(s) : 2 channels
Channel(s)_Original : 6 channels

For the record, QAAC isn't alone when it comes to the flags it writes. The Fraunhofer AAC encoders seem to do the same thing. fdkaac.exe and fhgaacenc.exe (the front end for the Fraunhofer AAC encoder which comes with Winamp).
NeroAAC is the odd one out. It's the only encoder for which the output file was reported as simply being 6 channel by MediaInfo.

After muxing with the video in an MKV though, MediaInfo just reports 6 channels for the audio regardless of the AAC encoder.

(fhgaacenc.exe sure is fast. At least 50% faster than the rest. VBR encoding).

Anyway, that's getting away from leon's problem a little, although he hasn't uploaded a sample......

Quote:
Originally Posted by leon View Post
Hi
MeGUI is good with 5.1 channel and it can detect channels ans channel positions.
with 2 channels most of the time can't detect positions.
and it's not good with mono but when I convert my audio to WAVE not matter 1,2 or 5.1 channel it works good.
I can't say I've ever experienced any channel count mismatches when converting stereo audio with MeGUI.

Last edited by hello_hello; 31st July 2014 at 23:57.
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Old 2nd August 2014, 00:13   #7619  |  Link
leon
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I said before that I converted the audio to WAVE with Xilisoft then encoded it so I don't have a sample.
wait hopefully I just found another file like that.
Code:
Audio
ID                                       : 2
Format                                   : AAC
Format/Info                              : Advanced Audio Codec
Format profile                           : LC
Codec ID                                 : 40
Duration                                 : 1h 3mn
Bit rate mode                            : Constant
Bit rate                                 : 56.0 Kbps
Channel(s)                               : 2 channels
Channel(s)_Original                      : 1 channel
Channel positions                        : Front: C
Sampling rate                            : 44.1 KHz
Compression mode                         : Lossy
Stream size                              : 25.5 MiB (10%)
the whole file is big so I can't upload it. is there anyway that I can upload the audio only untouched?

Last edited by leon; 2nd August 2014 at 00:17.
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Old 2nd August 2014, 09:24   #7620  |  Link
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You said the original audio is mono and the output audio is mono, and from what we've been able to tell so far "Channel(s) : 2 channels" as reported by MediaInfo is probably wrong and "Channel(s)_Original : 1 channel" is probably correct. MeGUI is letting you know via the log file that MediaInfo reports two channels, but according to the Avisynth output there's only one. Unfortunately MeGUI doesn't have an option to convert mono to stereo as such. Mind you I'm not sure if there's any advantage to doing so anyway.

You could try loading the MP4 directly into MeGUI's audio section and re-encoding the audio that way (it doesn't matter if it also contains video). Every time I do, MeGUI uses BassAudioSource(), and even if the audio is actually mono, the output ends up being stereo (I assume BassAudio decodes it to stereo?), which I gather is what you're wanting. Currently you're using FFAudioSource() which seems to decode mono as mono.
Hopefully with BassAudio you'll go from mono input to stereo output with no channel mismatch being reported at all. If you try it that way, please report back with the result.

There's possibly no point extracting the audio and uploading it, because if the channel mismatch is being caused by flags saved to the MP4 container, then after extracting the audio you'll probably just have a mono aac file. You could maybe try splitting off a small section of the whole MP4 with something like My MP4Box GUI for uploading. Or even try opening the MP4 with MKVMergeGUI and remuxing it as an MKV, then checking the MKV version with MediaInfo. If I'm correct, once it's remuxed as an MKV, MediaInfo will probably only report the audio as being mono, if that's what it is.

Last edited by hello_hello; 2nd August 2014 at 09:29.
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