VDM Won't Encode to MP3

[adalect]

Hi there.

So I just got a file in MKV format from the internet. I plan to make it playable on my DivX Player, so I have to convert it. I typically use XVID and MP3 Audio, so I stick to that.

So I make a script to import the MKV through AVISynth so VDM can read it.

Directshowsource("movie.mkv", fps=29.970, convertfps=true)
LanczosResize(720,400)

The ConvertFPS is to keep the video in sync, as I have previously learned over here. I also have to resize the video so that it's regular 16:9.

So I get the file readable in VDM. I go to the Audio Streams part of it to convert the raw audio I get into MP3. However, there is a problem: the audio formats to choose from in the Lame MP3 box is empty. A screenshot is at the bottom of the post.

So I thought it was nothing. I clicked "Show all formats" to override it and just choose what I choose often. When it comes time to encoding, VDM complains. It says the audio conversion is not possible. What happened?

[Baldrick]

What is the source audio?

Or maybe just try convert with something else like mkv2avi.

[adalect]

What is the source audio?

I believe it's AAC Audio, Two Channels (L and R). I am not quite sure though: I don't have the file on me right now. Will confirm in a couple of hours.

[adalect]

OK, here's the information that MediaInfo gave about the file.

General
Complete name                    : L:\Test\movie.mkv
Format                           : Matroska
File size                        : 72.1 MiB
Duration                         : 24mn 7s
Overall bit rate                 : 418 Kbps
Encoded date                     : UTC 2008-11-30 10:37:23
Writing application              : mkvmerge v2.0.2 ('You're My Flame') built on Feb 21 2007 23:40:55
Writing library                  : libebml v0.7.7 + libmatroska v0.8.1

Video
Format                           : AVC
Format/Info                      : Advanced Video Codec
Format profile                   : High@L5.1
Format settings, CABAC           : Yes
Format settings, ReFrames        : 8 frames
Codec ID                         : V_MPEG4/ISO/AVC
Duration                         : 24mn 6s
Nominal bit rate                 : 350 Kbps
Width                            : 720 pixels
Height                           : 480 pixels
Display aspect ratio             : 16/9
Frame rate                       : 29.970 fps
Standard                         : NTSC
Colorimetry                      : 4:2:0
Scan type                        : Progressive
Bits/(Pixel*Frame)               : 0.034
Writing library                  : x264 core 56 svn-676
Encoding settings                : cabac=1 / ref=5 / deblock=1:1:1 / analyse=0x3:0x133 / me=umh / subme=7 / brdo=1 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 / deadzone=21,11 / chroma_qp_offset=0 / threads=3 / nr=0 / decimate=1 / mbaff=0 / bframes=5 / b_pyramid=1 / b_adapt=1 / b_bias=0 / direct=1 / wpredb=1 / bime=1 / keyint=250 / keyint_min=25 / scenecut=40(pre) / rc=2pass / bitrate=350 / ratetol=1.0 / rceq='blurCplx^(1-qComp)' / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / pb_ratio=1.30

Audio
Format                           : AAC
Format/Info                      : Advanced Audio Codec
Format version                   : Version 4
Format profile                   : LC
Format settings, SBR             : No
Codec ID                         : A_AAC
Duration                         : 24mn 7s
Channel(s)                       : 2 channels
Channel positions                : L R
Sampling rate                    : 48.0 KHz
Resolution                       : 16 bits

So the Audio is AAC but it has two channels like regular MP3. Do you guys know what I can do from here? I think VDM or AVISynth may have something for me to use; I don't know what for sure though.

[adalect]

Another development: If I go to the Audio Conversion menu and change the Channels to "Stereo" from "No change (stereo)", the settings for LAME MP3 are available in the Audio Compression box. However, when I actually run the job, the program crashes. If I just click on "Save WAV" on the Audio Streams box to demux it with the conversion and compression settings, VDM also crashes.

What gives? I really don't know what to do now...

[Pinstripes23]

The method you're using now have you ever converted with aac audio as part of the file?

[DarrellS]

You can use Yamb to convert AVC/AAC MKV to a MP4 container which the newer versions of Virtualdub can open if you have the QT plugin and quicktime alternative.

The QT plugin can be found on the same page as Virtualdub. It will say MOV Virtualdub Plugin.

This can also be done with H264/AAC in an FLV container using FLVExtract to demux the files and Yamb to mux them into an MP4 container.

[adalect]

The method you're using now have you ever converted with aac audio as part of the file?

The method about "converting" certain properties of the sound like channels, and sample rate, are things I have not touched before.

As for converting AAC audio in general, I do not remember doing so specifically. What usually ends up happening is that the AVISynth file used to open any troublesome video gives me PCM (raw I think) audio, which I always convert to MP3 successfully.

It's just this time, even though the audio from the script is PCM; it won't convert. I believe it has something to do with it's AAC Source... but that source only has two channels two. It's an animation after all...

You can use Yamb to convert AVC/AAC MKV to a MP4 container

That won't help me one bit. I don't care about the container too much right now... I need to convert the audio to MP3 for the best compatibility with my DivX Player. And DarrellS, you sound a bit like a commercial... I don't think you intended that though.

[Pinstripes23]

The method you're using now have you ever converted with aac audio as part of the file?

The method about "converting" certain properties of the sound like channels, and sample rate, are things I have not touched before.

As for converting AAC audio in general, I do not remember doing so specifically. What usually ends up happening is that the AVISynth file used to open any troublesome video gives me PCM (raw I think) audio, which I always convert to MP3 successfully.

It's just this time, even though the audio from the script is PCM; it won't convert. I believe it has something to do with it's AAC Source... but that source only has two channels two. It's an animation after all...

That's why I was asking if you ever done aac before. I know avisynth should read the audio as PCM when using directshowsource. But you still need to decode the aac file. I have ffdshow do this for me. One way to test the avisynth script is play the file in a player like mpc and make sure the audio plays correctly with the video.

[DarrellS]

That won't help me one bit. I don't care about the container too much right now... I need to convert the audio to MP3 for the best compatibility with my DivX Player. And DarrellS, you sound a bit like a commercial... I don't think you intended that though.

Hey, we're just trying to help. You're technique seems to be working just fine though. The reason that I suggested using an MP4 container is because Virtualdub will read an MP4 container with the right plugin. VDM has not been upgraded in years.

I tried your script in Virtualdub 1.9.9 and had no problem at all though.

That's why I was asking if you ever done aac before. I know avisynth decompresses on-the-fly the audio so that vdubmod will read it as PCM. But you still need to decode the aac file. I have ffdshow do this for me. One way to test the avisynth script is play the file in a player like mpc and make sure the audio plays correctly with the video.

There is something wrong with the file that he downloaded. I hear that happens a lot with torrent downloads.

He could try to convert the AAC with an audio converter. I don't want to sound like a commercial but Nero Wave Editor works great for me. It's not free like the other programs that I mentioned but it does handle AAC very well.

[DarrellS]

Here's a guide for converting AAC to AC3 with MKVtoolnix and Videolan. Couldn't find a guide for MP3 but AC3 but your player should have no trouble with AC3.

http://www.kombitz.com/2009/02/24/how-to-convert-aac-to-ac3-in-mkv-file/

Not sure it will work with A_AAC since everything I've found has people having problems with this format. It seems to be different from AAC.

[Gavino]

One way to test the avisynth script is play the file in a player like mpc and make sure the audio plays correctly with the video.

At the same time, it's worth adding Info() to the script to see what audio properties Avisynth reports.

[Pinstripes23]

That's why I was asking if you ever done aac before. I know avisynth decompresses on-the-fly the audio so that vdubmod will read it as PCM. But you still need to decode the aac file. I have ffdshow do this for me. One way to test the avisynth script is play the file in a player like mpc and make sure the audio plays correctly with the video.

There is something wrong with the file that he downloaded. I hear that happens a lot with torrent downloads.

He could try to convert the AAC with an audio converter. I don't want to sound like a commercial but Nero Wave Editor works great for me. It's not free like the other programs that I mentioned but it does handle AAC very well.

Well that's why he should test the script cuz by the time it opens in vdubmod, the audio should be decompressed into PCM. But even then just decompress the audio to PCM wav separately. I don't know if you were suggesting that by using nero. Add it to another line in the avisynth script, then use the audiodub function to combine the video and audio.

[DarrellS]

Well that's why he should test the script cuz by the time it opens in vdubmod, the audio should be decompressed into PCM. But even then just decompress the audio to PCM wav separately. I don't know if you were suggesting that by using nero. Add it to another line in the avisynth script, then use the audiodub function to combine the video and audio.

No, I was recommending using Nero Wave Editor (or any other converter that would do the job) to convert the AAC to MP3 or AC3. I'm not sure that Nero would work anyway since after Googling last night, it seems a lot of people are having trouble with A_AAC audio which MediaInfo shows that he has in the file.

If I had a sample file, I could find out.

[adalect]

...avisynth should read the audio as PCM when using directshowsource. But you still need to decode the aac file. I have ffdshow do this for me. One way to test the avisynth script is play the file in a player like mpc and make sure the audio plays correctly with the video.

The original MKV file plays perfectly and the audio is perfectly fine. However, the audio from the script is awkward. I played the script through MPC as suggested, and the audio came out about 5x faster. I could tell because the "things that sounded like voices" were high pitched, and the theme song and its instruments started playing way too early. As well, if you skip the file from one part or another, the audio would stop completely.

My conclusion: something happened after the audio came from the MKV file and when it ran through the script.

Nero Wave Editor works great for me. It's not free

Sorry DarrellS, I am a bit cheap. The Nero Wave Editor is off the table: I am sure there is another cheaper way to do it.

it's worth adding Info() to the script to see what audio properties Avisynth reports.

The report is strange. A screenshot is at the bottom of the post.

Apparently, there are 6 audio channels even though MediaInfo reports two for the original file. MediaInfo says nothing for the script though. As I noted, there was no audio playing if you skipped through the file. The Script, though, picks up that there is some sort of audio. I am confused.


Thanks for are all your help so far. I am looking forward to solving this problem

[davexnet]

Have you tried the real Virtualdub, version 1.99, rather than the old and outdated vdubmod ?

[AlanHK]

The original MKV file plays perfectly and the audio is perfectly fine. However, the audio from the script is awkward. I played the script through MPC as suggested, and the audio came out about 5x faster. I could tell because the "things that sounded like voices" were high pitched, and the theme song and its instruments started playing way too early. As well, if you skip the file from one part or another, the audio would stop completely.

My conclusion: something happened after the audio came from the MKV file and when it ran through the script.

Try adding
EnsureVBRMP3sync()

That forces the audio to seek linearly. It slows down random access, so it takes a while to start playing if you jump through the file.

Also you can add
SoundOut()
(A plugin, you have to install, see SoundOut) which gives you an audio panel which can give info and also do conversions.)

If you really have more than 2 channels, you can use downmix functions in Avisynth to stereo.
See tebasuna's post

[adalect]

Um... I have another problem now. MPC says it can't render the audio from the script now. Before it gave the messed up audio, but now it refuses to spit out the script's audio all together. A screenshot is at the bottom of the post.

And here's the actual log:

L:\Test\MKV Files.avs::Avisynth audio #1
ACM Wrapper::Output

Media Type 0:
--------------------------
Audio: PCM 48000Hz 6ch 4608kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 12
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 6
nSamplesPerSec: 48000
nAvgBytesPerSec: 576000
nBlockAlign: 12
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 06 00 80 bb 00 00 00 ca 08 00 0c 00 10 00 ....€»...Ê......
0010: 00 00                                           ..


Media Type 1:
--------------------------
Audio: PCM 44100Hz stereo 1411kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 44100
nAvgBytesPerSec: 176400
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 44 ac 00 00 10 b1 02 00 04 00 10 00 ....D¬...±......
0010: 00 00                                           ..


Media Type 2:
--------------------------
Audio: PCM 44100Hz mono 705kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 44100
nAvgBytesPerSec: 88200
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 44 ac 00 00 88 58 01 00 02 00 10 00 ....D¬..ˆX......
0010: 00 00                                           ..


Media Type 3:
--------------------------
Audio: PCM 44100Hz stereo 705kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 44100
nAvgBytesPerSec: 88200
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 44 ac 00 00 88 58 01 00 02 00 08 00 ....D¬..ˆX......
0010: 00 00                                           ..


Media Type 4:
--------------------------
Audio: PCM 44100Hz mono 352kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 44100
nAvgBytesPerSec: 44100
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 44 ac 00 00 44 ac 00 00 01 00 08 00 ....D¬..D¬......
0010: 00 00                                           ..


Media Type 5:
--------------------------
Audio: PCM 22050Hz stereo 705kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 22050
nAvgBytesPerSec: 88200
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 22 56 00 00 88 58 01 00 04 00 10 00 ...."V..ˆX......
0010: 00 00                                           ..


Media Type 6:
--------------------------
Audio: PCM 22050Hz mono 352kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 22050
nAvgBytesPerSec: 44100
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 22 56 00 00 44 ac 00 00 02 00 10 00 ...."V..D¬......
0010: 00 00                                           ..


Media Type 7:
--------------------------
Audio: PCM 22050Hz stereo 352kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 22050
nAvgBytesPerSec: 44100
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 22 56 00 00 44 ac 00 00 02 00 08 00 ...."V..D¬......
0010: 00 00                                           ..


Media Type 8:
--------------------------
Audio: PCM 22050Hz mono 176kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 22050
nAvgBytesPerSec: 22050
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 22 56 00 00 22 56 00 00 01 00 08 00 ...."V.."V......
0010: 00 00                                           ..


Media Type 9:
--------------------------
Audio: PCM 11025Hz stereo 352kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 11025
nAvgBytesPerSec: 44100
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 11 2b 00 00 44 ac 00 00 04 00 10 00 .....+..D¬......
0010: 00 00                                           ..


Media Type 10:
--------------------------
Audio: PCM 11025Hz mono 176kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 11025
nAvgBytesPerSec: 22050
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 11 2b 00 00 22 56 00 00 02 00 10 00 .....+.."V......
0010: 00 00                                           ..


Media Type 11:
--------------------------
Audio: PCM 11025Hz stereo 176kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 11025
nAvgBytesPerSec: 22050
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 11 2b 00 00 22 56 00 00 02 00 08 00 .....+.."V......
0010: 00 00                                           ..


Media Type 12:
--------------------------
Audio: PCM 11025Hz mono 88kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 11025
nAvgBytesPerSec: 11025
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 11 2b 00 00 11 2b 00 00 01 00 08 00 .....+...+......
0010: 00 00                                           ..


Media Type 13:
--------------------------
Audio: PCM 8000Hz stereo 256kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 8000
nAvgBytesPerSec: 32000
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 40 1f 00 00 00 7d 00 00 04 00 10 00 ....@....}......
0010: 00 00                                           ..


Media Type 14:
--------------------------
Audio: PCM 8000Hz mono 128kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 8000
nAvgBytesPerSec: 16000
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 40 1f 00 00 80 3e 00 00 02 00 10 00 ....@...€>......
0010: 00 00                                           ..


Media Type 15:
--------------------------
Audio: PCM 8000Hz stereo 128kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 8000
nAvgBytesPerSec: 16000
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 40 1f 00 00 80 3e 00 00 02 00 08 00 ....@...€>......
0010: 00 00                                           ..


Media Type 16:
--------------------------
Audio: PCM 8000Hz mono 64kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 8000
nAvgBytesPerSec: 8000
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 40 1f 00 00 40 1f 00 00 01 00 08 00 ....@...@.......
0010: 00 00                                           ..


Media Type 17:
--------------------------
Audio: PCM 48000Hz stereo 1536kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 48000
nAvgBytesPerSec: 192000
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 80 bb 00 00 00 ee 02 00 04 00 10 00 ....€»...î......
0010: 00 00                                           ..


Media Type 18:
--------------------------
Audio: PCM 48000Hz mono 768kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 48000
nAvgBytesPerSec: 96000
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 80 bb 00 00 00 77 01 00 02 00 10 00 ....€»...w......
0010: 00 00                                           ..


Media Type 19:
--------------------------
Audio: PCM 48000Hz stereo 768kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 48000
nAvgBytesPerSec: 96000
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 80 bb 00 00 00 77 01 00 02 00 08 00 ....€»...w......
0010: 00 00                                           ..


Media Type 20:
--------------------------
Audio: PCM 48000Hz mono 384kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 48000
nAvgBytesPerSec: 48000
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 80 bb 00 00 80 bb 00 00 01 00 08 00 ....€»..€»......
0010: 00 00                                           ..


Media Type 21:
--------------------------
Audio: PCM 32000Hz stereo 1024kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 4
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 32000
nAvgBytesPerSec: 128000
nBlockAlign: 4
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 00 7d 00 00 00 f4 01 00 04 00 10 00 .....}...ô......
0010: 00 00                                           ..


Media Type 22:
--------------------------
Audio: PCM 32000Hz mono 512kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 32000
nAvgBytesPerSec: 64000
nBlockAlign: 2
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 00 7d 00 00 00 fa 00 00 02 00 10 00 .....}...ú......
0010: 00 00                                           ..


Media Type 23:
--------------------------
Audio: PCM 32000Hz stereo 512kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 2
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 2
nSamplesPerSec: 32000
nAvgBytesPerSec: 64000
nBlockAlign: 2
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 02 00 00 7d 00 00 00 fa 00 00 02 00 08 00 .....}...ú......
0010: 00 00                                           ..


Media Type 24:
--------------------------
Audio: PCM 32000Hz mono 256kbps

AM_MEDIA_TYPE: 
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 1
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x0001
nChannels: 1
nSamplesPerSec: 32000
nAvgBytesPerSec: 32000
nBlockAlign: 1
wBitsPerSample: 8
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 00 01 00 00 7d 00 00 00 7d 00 00 01 00 08 00 .....}...}......
0010: 00 00                                           ..

It didn't give me this warning before. I restarted and it still gave me this stuff. The only new thing I tried was AlanHK's "EnsureVBRMP3sync()" command, but that froze on me. The next time I ran the script throguh MPC, it gave me this.

I also tried the SoundOut command. The conversion it spat out was bout 2x slower than the original speed. If I speed it up in MPC, it sounds OK but it's still too fast for the video.

Sorry for the rushed reply: I am feeling a bit sick now.

[DarrellS]

Sorry DarrellS, I am a bit cheap. The Nero Wave Editor is off the table: I am sure there is another cheaper way to do it.

I searched around and this is what I came up with...

Audacity 1.3.12 Beta with the FFmpeg Import/Export Library and the Lame MP3 Encoder.

Sorceforge is down right now so you'll have to Google the Audacity download (EDIT: just click on audacity to take you to the download page and right click, save as) but the plugins can be found on this page...

http://manual.audacityteam.org/index.php?title=FAQ:Installation_and_Plug-Ins#installffmpeg

...with instructions on how to install them.

Under Preferences > Import/Export, you'll need to change "when exporting tracks to an audio file" to "use custom mix" or it will save MP3 as mono.

Hope this works on your file but I have a bad feeling that it won't. When I was reading last night, people were saying that the header was messed up on A_AAC files so you may have to try and find a way to fix the header.

Good luck!


BTW, all of your attachments say "invalid link".

[DarrellS]

You could also try Foobar2000. You'll need to direct it to Lame.exe to encode to MP3.

[adalect]

Audacity 1.3.12 Beta with the FFmpeg Import/Export Library and the Lame MP3 Encoder.

I already extracted the pure AAC audio with MKVExtract, so I ran that through Audacity. Didn't work. Its diagram even indicated the length of the audio was off by a third of its original length. I don't know... I feel this problem isn't solvable.

all of your attachments say "invalid link".

When I log in, they show up fine. I think you have to log in before it works.

[adalect]

If I had a sample file, I could find out.

http://www.megaupload.com/?d=LK488RI1

Tell me what you figure out. I sort of give up... but I want some closure in this thread for other guys tackling the same problem.

[Pinstripes23]

You could try loading the mkv clip into avisynth using ffmpegsource - http://code.google.com/p/ffmpegsource/downloads/list
Download and extract the files into your avisynth plugins folder, then use the script:

ffmpegsource2("file.mkv", atrack=-1)

And open the script in mpc, but you have to wait like a min or 2 depending how long your file gets indexed. I use this way as a secondary resort sometimes.

Oh BTW I also can't see your attachments and I'm logged in

[Pinstripes23]

If I had a sample file, I could find out.

http://www.megaupload.com/?d=LK488RI1

Tell me what you figure out. I sort of give up... but I want some closure in this thread for other guys tackling the same problem.

Is that the whole file? Directshowsource opens both audio and video for me with no problems and plays in mpc no problems. Your system is not decoding it properly for some reason, so the problem is not the file itself. I don't think you ever told me how you decode aac...

[adalect]

Your system is not decoding it properly for some reason, so the problem is not the file itself. I don't think you ever told me how you decode aac...

I don't know myself either. I am using the K-Lite Mega Codec Pack though...

[Pinstripes23]

Try using ffdshow instead cuz it seems to work on this particular audio file.

Also if you are unsure how a file is decoded open the file in gspot and under "proposed codec solutions and test" you can see how your system is decoding it.

[DarrellS]

I have no problem using Virtualdub 1.9.9 with the Avisynth script in the first post.

Audacity creates the MP3 from the AAC file but it plays in slow motion.

Converting to MP4 and trying to compress to MP3 causes Virtualdub to crash.

Foobar converts the AAC file to MP3 with no problems.


I don't use codec packs at all. I use only the codecs that I need and download them from the builders website.

I do use ffdshow but I only use VFW codecs and only for special file types that Virtualdub needs to open the files.

I have PowerDVD 7, Nero 7 ultra and DivX Plus which has H264 and AAC decoders (DivX H264 and DivX AAC is what is listed under GSpot to open the file and looks like what the file was created with). DivX Plus is free.

[adalect]

You could try loading the mkv clip into avisynth using ffmpegsource - http://code.google.com/p/ffmpegsource/downloads/list
Download and extract the files into your avisynth plugins folder, then use the script:

ffmpegsource2("file.mkv", atrack=-1)

And open the script in mpc, but you have to wait like a min or 2 depending how long your file gets indexed.

This works! But I have two questions:

1. What is this "movie.mkv.ffindex" file that gets generated when I run the script?
2. If I use the FFMpegSource2 Command, will the audio sync be off? I heard that MKV usually has variable framerates or something like that... that is why I used the convertfps thing on DirectShowSource.

[AlanHK]

What is this "movie.mkv.ffindex" file that gets generated when I run the script?

What it sounds like: an index used by ffmpegsource2.
It will recreate it if you delete it.
If it's there and up to date, fmpegsource2 will start faster.

If I use the FFMpegSource2 Command, will the audio sync be off? I heard that MKV usually has variable framerates or something like that... that is why I used the convertfps thing on DirectShowSource.

I think that's one thing the index does. Usually sync is not a problem with ffmpegsource2; but always check; there can be errors or inherent delays in the file that you have to adjust for.

But see this, though it may not be relevant as it refers to the older version of ffmpegsource.

Just play a few random bits and confirm that it's in sync.

[Pinstripes23]

This works! But I have two questions:

1. What is this "movie.mkv.ffindex" file that gets generated when I run the script?
2. If I use the FFMpegSource2 Command, will the audio sync be off? I heard that MKV usually has variable framerates or something like that... that is why I used the convertfps thing on DirectShowSource.

That's good you got it to work for you. Those 2 questions have pretty much been answered already but to also add as far as sync issues, the only time I experienced them were on Mpeg2 HD w/ multi channel AC3. Other containers I've used it on such as avi, mkv, mp4, etc....I didn't experience any sync issues. But like I said I only really use it as a secondary means, if directshowsource/avisource fails.