Converting to AAC 5.1 / 7.1 with ffmpeg changes channel layout

[vash1]

Hi guys,

When converting DTS HD-MA or multichannel FLAC to AAC with the latest ffmpeg 4.1.3 I get channel layout that mismatch source audio file.

This happens for 5.1 and 7.1 audios.

The command I use is the simplest one:

ffmpeg -i 7.1.flac 7.1.aac

ffmpeg -i 5.1.dts 5.1.aac

The source and destination layouts looks like the following:

L R C LFE Lb Rb Ls Rs <-- 7.1.flac
C L R Ls Rs Lb Rb LFE <-- 7.1.aac

C L R Ls Rs LFE <-- 5.1.dts
L R C Cb Lb Rb <-- 5.1.aac

Please kindly assist.

[pandy]

Sounds like serious regression for ffmpeg.
Try to restore proper order with https://ffmpeg.org/ffmpeg-filters.html#channelmap
Check channel layout for source and for target.

[eddy89]

Any update here? Does anyone know how to fix it?
I have DTS-HD MA 7.1:

Channel layout                           : C L R LFE Lb Rb Lss Rss

And after converting to aac:

Channel layout                           : C L R Ls Rs Lw Rw LFE

Is it fine, is incorrect? By default, if I was not specifying 8 channels in the command line, it was just taking 7 in the output of the aac. Is kind of weird that this is not autodetected by default.

[ProWo]

The channel layout for DTS to aac 8ch is correct with ffmpeg v7.1.1

[eddy89]

Thanks for the reply.

Can you develop on what is the meaning of the Lb Rb Lss Rss as opposed to Ls Rs Lw Rw?

DTS:

Channel layout                           : C L R LFE Lb Rb Lss Rss

AAC:

Channel layout                           : C L R Ls Rs Lw Rw LFE

Apart from that, do you know why unless I add this to the command line:

-ac 8

to make sure I am outputting 8 channels, it just creates a file with just 7 channels? This was before the previous flag in command line:

Channel(s)                               : 7 channels
Channel layout                           : C L R Ls Rs Lb Rb

For me it seems quite concerning, as I guess knowing the number of input/output channels, should be completely automatic (unless you want to explicitly change it so), right?

Besides that, see the difference in the channels, apart from the lack of one of them, the naming is quite different:

Channel(s)                               : 7 channels
Channel layout                           : C L R Ls Rs Lb Rb
Channel(s)                               : 8 channels
Channel layout                           : C L R Ls Rs Lw Rw LFE

and the only difference is just the "-ac 8" flag.

Hope anyone can put some light here.

Thanks

[ProWo]

Can you develop on what is the meaning of the Lb Rb Lss Rss as opposed to Ls Rs Lw Rw?

DTS:

Channel layout                           : C L R LFE Lb Rb Lss Rss

AAC:

Channel layout                           : C L R Ls Rs Lw Rw LFE

Which software gives you these channel names?

Apart from that, do you know why unless I add this to the command line:

-ac 8

to make sure I am outputting 8 channels, it just creates a file with just 7 channels?

Post your full ffmpeg commandline that you use for the DTS to AAC conversion.

[eddy89]

Can you develop on what is the meaning of the Lb Rb Lss Rss as opposed to Ls Rs Lw Rw?

DTS:

Channel layout                           : C L R LFE Lb Rb Lss Rss

AAC:

Channel layout                           : C L R Ls Rs Lw Rw LFE

Which software gives you these channel names?

MediaInfo.

Apart from that, do you know why unless I add this to the command line:

-ac 8

to make sure I am outputting 8 channels, it just creates a file with just 7 channels?

Post your full ffmpeg commandline that you use for the DTS to AAC conversion.

ffmpeg -i /path/audio DTS-HD\ MA\ 7.1\ @\ 2649\ kbps.ENG.dts -c aac_at -aac_at_quality 0 mynewaudio.m4a

I was using the flag "-ac 8" to be able to get the 8 channels, otherwise I would get 7.

[ProWo]

ffmpeg -i /path/audio DTS-HD\ MA\ 7.1\ @\ 2649\ kbps.ENG.dts -c aac_at -aac_at_quality 0 mynewaudio.m4a

I was using the flag "-ac 8" to be able to get the 8 channels, otherwise I would get 7.

Maybe the audio codec (aac_at, unknown to me and my latest nightly ffmpeg version).

[cholla]

Maybe the audio codec (aac_at, unknown to me and my latest nightly ffmpeg version).

I was curious about this & searched it.
It is an Apple codec.

https://trac.ffmpeg.org/wiki/Encode/AAC


At the bottom of this web page under ffmpeg it has some limited information about using with ffmpeg build:
https://wiki.hydrogenaud.io/index.php?title=Apple_AAC

[pandy]

I vaguely recall that Apple may use different default channel allocation in their AAC - ffmpeg may fail in autodetection - can't provide you link but im almost sure that this was on hydrogen forum. I would try to use channel remap filter on ffmpeg to restore proper channel allocation - perhaps configuring filter 'channelmap' in ffmpeg in proper way will improve this (signaling proper channel layout) - https://ffmpeg.org/ffmpeg-filters.html#channelmap ; https://trac.ffmpeg.org/wiki/AudioChannelManipulation#Remapchannels .
Btw ffmpeg recognize four different 7.1 channel layouts - not sure how many of them is supported by 'aac_at' codec - this can be also important.

[eddy89]

I vaguely recall that Apple may use different default channel allocation in their AAC - ffmpeg may fail in autodetection - can't provide you link but im almost sure that this was on hydrogen forum. I would try to use channel remap filter on ffmpeg to restore proper channel allocation - perhaps configuring filter 'channelmap' in ffmpeg in proper way will improve this (signaling proper channel layout) - https://ffmpeg.org/ffmpeg-filters.html#channelmap ; https://trac.ffmpeg.org/wiki/AudioChannelManipulation#Remapchannels .
Btw ffmpeg recognize four different 7.1 channel layouts - not sure how many of them is supported by 'aac_at' codec - this can be also important.

I tried (before) the same with the default aac before apple (which I use because is supposed to be the one with better quality), and the channel mapping is the same as the ffmpeg default aac encoder.

[ProWo]

I tried (before) the same with the default aac before apple (which I use because is supposed to be the one with better quality), and the channel mapping is the same as the ffmpeg default aac encoder.

I have just tried this with the normal aac encoder from ffmpeg.
I then loaded the dts and the aac file into Audacity at the same time and checked the individual tracks; the channel layout is correct, the individual aac channels are exactly the same as in the dts file.

[pandy]

I tried (before) the same with the default aac before apple (which I use because is supposed to be the one with better quality), and the channel mapping is the same as the ffmpeg default aac encoder.

You need to be sure that you are using proper channel layout - not sure if ffmpeg is able to automatically remap different channel layouts and depends on information source there are at least 3 different 7.1 channel layouts:

7.1            FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide)      FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side) FL+FR+FC+LFE+FLC+FRC+SL+SR

or based on other source 4 layouts:

7.1            FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide)      FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side) FL+FR+FC+LFE+FLC+FRC+SL+SR
7.1(top)       FL+FR+FC+LFE+BL+BR+TFL+TFR

There must be match between source channel layout and destination supported channel layout - if target channel layout is different than source then perhaps you need to perform some downmixing - channel mixing operation accordingly to standard (unknown to me in case of layouts higher than 5.1).

[eddy89]

Thanks pandy, I don't want to do any downmixing at all.
Wanna preserve all the channels.

Hope anyone can help me with encode DTS HD-MA 7.1 to AAC 7.1 (without losing any channel), here: https://forum.videohelp.com/threads/417937-Audio-DTS-conversion#post2772491

[ProWo]

Hope anyone can help me with encode DTS HD-MA 7.1 to AAC 7.1 (without losing any channel)

This works for me:

FFmpeg.exe  -i "input_dts_ma_8ch.dts"  -c:a aac -b:a 512k -ar 48000  "output.aac"

The output is aac 8ch.

[eddy89]

Hope anyone can help me with encode DTS HD-MA 7.1 to AAC 7.1 (without losing any channel)

This works for me:

FFmpeg.exe  -i "input_dts_ma_8ch.dts"  -c:a aac -b:a 512k -ar 48000  "output.aac"

The output is aac 8ch.

I am trying to replicate the configuration (with less bitrate cause is insane at almost 900k) of a given release group movie (tigole), which has a AAC 7.1, allegedly properly done.
Their configuration when I have a look in media info is this:

Audio
ID : 2
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Codec ID : A_AAC-2
Duration : 1 h 53 min
Bit rate : 897 kb/s
Channel(s) : 8 channels
Channel layout : C L R Ls Rs Lb Rb LFE
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Delay relative to video : 20 ms
Stream size : 728 MiB (5%)
Language : English
Default : Yes
Forced : No

And mine (with your command line), is this:

General
Complete name : /Users/legolas/Downloads/output.aac
Format : ADTS
Format/Info : Audio Data Transport Stream
File size : 418 MiB
Overall bit rate mode : Variable

Audio
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Format version : Version 4
Codec ID : 2
Bit rate mode : Variable
Channel(s) : 8 channels
Channel layout : C L R Ls Rs Lw Rw LFE
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Stream size : 418 MiB (100%)

See how the channel layout is different! Lb Rb vs Lw Rw?? I don't understand what it means...

[cholla]

@ eddy89 ,
Can you post a clip of the original audio you are working with ?
ffmpeg can create clips & so can AviDemux.
Other software can as well.
The clip can be up to 500MB for most formats.
A smaller clip would be better though .
Just not too small.

[cholla]

I downloaded a file something like eddy89 has.
This is the MediaInfo on this file with the video removed:

Audio
ID                                       : 1
Format                                   : DTS XLL
Format/Info                              : Digital Theater Systems
Commercial name                          : DTS-HD Master Audio
Codec ID                                 : A_DTS
Duration                                 : 49 s 54 ms
Bit rate mode                            : Variable
Channel(s)                               : 8 channels
Channel layout                           : C L R LFE Lb Rb Lss Rss
Sampling rate                            : 96.0 kHz
Frame rate                               : 187.500 FPS (512 SPF)
Bit depth                                : 24 bits
Compression mode                         : Lossless
Language                                 : English
Default                                  : Yes
Forced                                   : No

This is the file using -c:a libfdk_aac in ffmpeg:

Audio
ID                                       : 1
Format                                   : AAC LC
Format/Info                              : Advanced Audio Codec Low Complexity
Codec ID                                 : mp4a-40-2
Duration                                 : 49 s 56 ms
Source duration                          : 49 s 76 ms
Bit rate mode                            : Constant
Bit rate                                 : 1 257 kb/s
Channel(s)                               : 8 channels
Channel layout                           : C L R Ls Rs Lb Rb LFE
Sampling rate                            : 96.0 kHz
Frame rate                               : 93.750 FPS (1024 SPF)
Compression mode                         : Lossy
Stream size                              : 7.35 MiB (100%)
Source stream size                       : 7.35 MiB (100%)
Language                                 : English
Default                                  : Yes
Alternate group                          : 1

The acronyms in Channel layout are different but I do not have any that are rw or lw.
These are the errors I get when I use -c:a aac_at in ffmpeg:
It appears that the Apple aac_at does not work with DTS-HD MA 7.1

[aac_at @ 006d8940] AudioToolbox init error: 1718449215
[aost#0:0/aac_at @ 007114c0] [enc:aac_at @ 006d88c0] Error while opening encoder - maybe incorrect parameters such as bit_rate, rate, width or height.
[af#0:0 @ 006d9640] Error sending frames to consumers: Unknown error occurred
[af#0:0 @ 006d9640] Task finished with error code: -1313558101 (Unknown error occurred)
[af#0:0 @ 006d9640] Terminating thread with return code -1313558101 (Unknown error occurred)
[aost#0:0/aac_at @ 007114c0] [enc:aac_at @ 006d88c0] Could not open encoder before EOF
[aost#0:0/aac_at @ 007114c0] Task finished with error code: -22 (Invalid argument)
[aost#0:0/aac_at @ 007114c0] Terminating thread with return code -22 (Invalid argument)
[out#0/ipod @ 006d8340] Nothing was written into output file, because at least one of its streams received no packets.

Using ProWo's code MediaInfo:

Audio
Format                                   : AAC LC
Format/Info                              : Advanced Audio Codec Low Complexity
Format version                           : Version 4
Codec ID                                 : 2
Bit rate mode                            : Variable
Channel(s)                               : 8 channels
Channel layout                           : C L R Ls Rs Lw Rw LFE
Sampling rate                            : 48.0 kHz
Frame rate                               : 46.875 FPS (1024 SPF)
Compression mode                         : Lossy
Stream size                              : 2.76 MiB (100%)

Prowo's code with extension changed to .m4a:

Audio
ID                                       : 1
Format                                   : AAC LC
Format/Info                              : Advanced Audio Codec Low Complexity
Codec ID                                 : mp4a-40-2
Duration                                 : 49 s 56 ms
Source duration                          : 49 s 76 ms
Bit rate mode                            : Variable
Bit rate                                 : 469 kb/s
Maximum bit rate                         : 512 kb/s
Channel(s)                               : 8 channels
Channel layout                           : C L R Ls Rs Lw Rw LFE
Sampling rate                            : 48.0 kHz
Frame rate                               : 46.875 FPS (1024 SPF)
Compression mode                         : Lossy
Stream size                              : 2.74 MiB (99%)
Source stream size                       : 2.74 MiB (99%)
Language                                 : English
Default                                  : Yes
Alternate group                          : 1

It appears that the ffmpeg Native acc codec uses the lw & rw for the back surround speakers.

[eddy89]

@ eddy89 ,
Can you post a clip of the original audio you are working with ?
ffmpeg can create clips & so can AviDemux.
Other software can as well.
The clip can be up to 500MB for most formats.
A smaller clip would be better though .
Just not too small.

Find it here (sorry, they are the full files, not clips):

my compression with ProWo's command line.aac:
https://1fichier.com/?8ibyukvj2yooofcqdjaj

audio demuxed with mkvtoolnix from tigole group release.aac:
https://1fichier.com/?cuw2q9o8ih7j01zwfoxc

alleged original DTS 7.1 from another release.dts:
https://1fichier.com/?7c8oc5zpqx1ux3w9cd5m

Please have a look.

It appears that the ffmpeg Native acc codec uses the lw & rw for the back surround speakers.

I don't know the meaning of lw and rw... Don't understand the difference. Can you elaborate and explain?

It appears that the Apple aac_at does not work with DTS-HD MA 7.1

Weird, mine seems to work, the only problem is that it looses a channel.

Input #0, dts, from '...whatever.dts':
  Duration: N/A, start: 0.000000, bitrate: N/A
  Stream #0:0: Audio: dts (dca) (DTS-HD MA), 48000 Hz, 7.1, s32p (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (dts (dca) -> aac (aac_at))
Press [q] to stop, [?] for help
Output #0, adts, to '/Users/blah/Downloads/output2b.aac':
  Metadata:
    encoder         : Lavf61.7.100
  Stream #0:0: Audio: aac, 48000 Hz, 7.0, s16, 512 kb/s
      Metadata:
        encoder         : Lavc61.19.101 aac_at
[out#0/adts @ 0x7fd3dd906e00] video:0KiB audio:425472KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 0.512698%
size=  427653KiB time=01:53:27.48 bitrate= 514.6kbits/s speed=23.4x

It looses a channel, but at least does not error out... (ProWo's command line, just adding the aac_at codec), although it seems quite dodgy being able to detect 8 channels but outputting 7.

This is the file using -c:a libfdk_aac in ffmpeg:

Just installed the libfdk_aac, and when I use it, the channel layout is what you said: it is not the same as the default aac.

Related with that, when I use the "libfdk_aac" instead of the normal "aac", with ProWo's command line, the bitrate is not variable anymore, maybe the default configuration with the libfdk_aac is different than the standard?

Thank you very much

[cholla]

I don't know the meaning of lw and rw... Don't understand the difference. Can you elaborate and explain?

I'm not sure eother.I believe that lw and rw are left wide & right wide.
I believe this is the same as lb & rb. Left back & Right back.
I believe on an amp these would come from the same jacks.
The positioning of the speakers might be a little different for the best listening.
An object-based format instead of an area based format.
Maybe a more knowledgeable member can explain this better.

Weird, mine seems to work, the only problem is that it looses a channel.
It looses a channel, but at least does not error out... (ProWo's command line, just adding the aac_at codec), although it seems quite dodgy being able to detect 8 channels but outputting 7.

It may be the sample file I'm using. I just got one from the internet.
I will see how it works when I download the last file you posted.
The site requires an hour wait between downloads.

Just installed the libfdk_aac, and when I use it, the channel layout is what you said: it is not the same as the default aac.

Related with that, when I use the "libfdk_aac" instead of the normal "aac", with ProWo's command line, the bitrate is not variable anymore, maybe the default configuration with the libfdk_aac is different than the standard?

This is a variation of ProWo's code:Give it a try

ffmpeg -i input_dts_ma_8ch.dts -c:a libfdk_aac -vbr 5 output.aac
OR
ffmpeg -i input_dts_ma_8ch.dts -c:a libfdk_aac -vbr 5 output.m4a

I will post the results when I download the last file.