I should add that this is a workaround that I *think* could be unnecessary if WD did a firmware update whereby AAC could be output as PCM to the receiver. Not 100% sure about this though.
Conversion to multichannel PCM would only work for HDMI receivers. AFAIK, optical doesn’t have the bandwidth for 6 uncompressed channels of audio. Not sure if any other NMT manufacturers using the same Sigma chip offer this feature… last I read, even Poporn Hour users have to recode AAC 5.1 tracks to get them working with AVRs.
On the plus side, Heartware’s Audio Converter works well, not just for AAC 5.1 but for DTS too. I’ve been able to recode DTS tracks into AC3 and output a new MKV with both tracks really easily. When I finally upgrade my AVR, I can still use DTS.
I am not able to make Heartware’s Audio Converter work. It always say that source file in use which is not correct. Any other audio encoder?
Actually, the best software to re-encode almost anything (video or audio) is ffmpeg. This freeware is at the heart of most of nearly all freeware GUIs (and many commercial products) you can get. It’s command line driven, but that shouldn’t scare you away as there are plenty of example tutorials out there to help you with the syntax.
Yeah i am familiar with command line interface, but i checked the docs and I dont see any command which transcodes the audio.
Do you know any?
It will be helpfull for lot of people who are struggling to play multi channel sounds in wd live.
It all depends on what you want to do. I haven’t looked back at this entire thread, so you’ll need to tell me exactly what you need the audio to look like.
ffmpeg -i test.mkv -acodec mp2 outfile.mkv
will, for example, change the audio codec to MPEG-2. Perhaps these sites will be helpful:
I want to convert audio of my mkv file from AAC to stereo. Here is the information of audio in my mkv file:
======
ID : 2 Format : AAC Format/Info : Advanced Audio Codec Format version : Version 4 Format profile : LC Format settings, SBR : Yes Format settings, PS : No Codec ID : A_AAC Duration : 2h 20mn Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz
I tried it but it gives me an error. I have the latest version
===============
E:\WDLive\FFmpeg-svn-22140>ffmpeg -i wus.mkv -acodec aac -ac 2 wus2.mkv
FFmpeg version SVN-r22140-Sherpya, Copyright (c) 2000-2010 the FFmpeg developers
built on Mar 2 2010 03:26:44 with gcc 4.2.5 20080919 (prerelease) [Sherpya]
libavutil 50. 9. 0 / 50. 9. 0
libavcodec 52.55. 0 / 52.55. 0
libavformat 52.54. 0 / 52.54. 0
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.17. 0 / 1.17. 0
libswscale 0.10. 0 / 0.10. 0
libpostproc 51. 2. 0 / 51. 2. 0
[matroska @ 0190d060]Estimating duration from bitrate, this may be inaccurate
Seems stream 0 codec frame rate differs from container frame rate: 47.95 (5994/1
25) → 24.00 (24/1)
Input #0, matroska, from ‘wus.mkv’:
Duration: 02:20:00.72, start: 0.000000, bitrate: N/A
Chapter #0.0: start 0.097000, end 8400.726000
Metadata:
title : 00:00:00.097
Stream #0.0(eng): Video: h264, yuv420p, 640x272, PAR 1:1 DAR 40:17, 47.62 fp
s, 24 tbr, 1k tbn, 47.95 tbc
Stream #0.1(hin): Audio: aac, 24000 Hz, 5.1, s16
Output #0, matroska, to ‘wus2.mkv’:
Metadata:
encoder : Lavf52.54.0
Stream #0.0(eng): Video: mpeg4, yuv420p, 640x272 [PAR 1:1 DAR 40:17], q=2-31
, 200 kb/s, 1k tbn, 24 tbc
Stream #0.1(hin): Audio: aac, 24000 Hz, 2 channels, s16, 64 kb/s
Stream mapping:
Stream #0.0 → #0.0
Stream #0.1 → #0.1
Press [q] to stop encoding
[aac @ 01a53940]SBR not implemented. Update your FFmpeg version to the newest on
e from SVN. If the problem still occurs, it means that your file has a feature w
hich has not been implemented.
[aac @ 01a53940]SBR not implemented. Update your FFmpeg version to the newest on
e from SVN. If the problem still occurs, it means that your file has a feature w
hich has not been implemented.
[aac @ 01a53940]SBR not implemented. Update your FFmpeg version to the newest on
e from SVN. If the problem still occurs, it means that your file has a feature w
hich has not been implemented.
Resampling with input channels greater than 2 unsupported.
Can not resample 6 channels @ 24000 Hz to 2 channels @ 24000 Hz
No, it’s the multi-channel thing. There is a lot of talk about this on the web – apparently a lot of folks would like to downmix using FFMPEG – but the only workarounds involve some heavy scripting.
If it were me I’d use Sound Forge Pro but that’s what I have. I understand Audacity is pretty close, though (and isn’t it freeware?)
I have added on the “ideas” section the suggestion to have the AAC 5.1 converted on the fly to DolbyDigital 5.1. But nobody cared to vote for it. Anyway, because of the incompatibility with SPDIF and current receivers, I stay away from AAC files.
It supports it in the sense that it will pass through the AC3 or DTS to a receiver that can decode it (when “Digital” is selected in the Audio/Video options). But it’s only AC3 or DTS 5.1 that is supported.
When you select “Stereo” it will convert the audio to stereo.