Encoding AC3 to AAC with NeroAAC Encoder

[El Heggunte]

.............

If you really mean VBR (variable bit rate), then that's not an issue either for playback. Only a possible issue when editing (you also have the option of using CBR for AAC, IIRC some encoders like FAAC only allow CBR)

You recall incorrectly , it's enc_aacplus that cannot do VBR:

[C:\]
=>aac+encoder

********************************************************************
* AACPlus v2 Encoder (using Winamp enc_aacplus.dll)
* Coding Technologies encoder 8.0.3
* Build Dec 29 2009, 08:04:23
********************************************************************

Usage:
        aac+encoder <wav_file> <bitstream_file> [options]
Options:
        --br       - Set bitrate (CBR) to <bitrate> bps. Default is 128000
        --mono     - Encode as Mono
        --ps       - Enable Parametric Stereo (bitrates up to 56000)
        --is       - Independent Stereo, disables Joint Stereo M/S coding
        --dc       - Prefer Dual Channels
        --he       - Encode as HE-AAC (bitrate up to 128000/213335)
        --lc       - Encode as LC-AAC (bitrate up to 320000)
        --high     - Encode as HE-AAC with High Bitrates
                    (bitrates up to 256000, multichannel is not supported)
        --speech   - Tune for Speech
        --pns      - Enable Perceptual Noise Subsitution (PNS)
        --mpeg2aac - Force MPEG2 AAC stream
        --mpeg4aac - Force MPEG4 AAC stream
        --rawpcm    <rate> <cnt> <bp> - Signal RAW PCM input instead of WAV
                    <rate> - Samplerate in Hz (32000, 44100 or 48000)
                    <cnt>  - Channels count (1 or 2)
                    <bp>   - Bit's per sample (8 or 16)
WARNING: this encoder can read and encode data from stdin:
         use - as input filename
The output can be a raw .aac file or a MPEG4 ISO compilant .mp4/.m4a
(libmp4v2.dll [from Winamp folder] must be in the same directory)

As for faac,

[C:\]
=>faac --long-help
Freeware Advanced Audio Coder
FAAC 1.28

Usage: faac [options] infiles ...

Quality-related options:
  -q <quality>  Set default variable bitrate (VBR) quantizer quality in percent.
                (default: 100, averages at approx. 120 kbps VBR for a normal
                stereo input file with 16 bit and 44.1 kHz sample rate; max.
                value 500, min. 10).
  -b <bitrate>  Set average bitrate (ABR) to approximately <bitrate> kbps.
                (max. value 152 kbps/stereo with a 16 kHz cutoff, can be raised
                with a higher -c setting).
  -c <freq>     Set the bandwidth in Hz (default: automatic, i.e. adapts
                maximum value to input sample rate).
<etc Etc ETC>

[Selur]

btw. this is an example how to use qaac to encode to alac:

ffmpeg -y -analyzeduration 100M -probesize 100M -threads 2 -v -10 -i "H:\Temp\18_28_34_1010_01_iId_1_aid_1.ac3" -ac 6 -acodec pcm_s16le -f wav - | qaac --no-delay --threading --raw-channels 6 --raw-rate 48000 --alac --raw - -o "H:\Temp\iId_1_aid_1_18_28_34_1010_02.aac"

[poisondeathray]

You recall incorrectly , it's enc_aacplus that cannot do VBR:

I think CT-AAC is CBR only too ? Not sure can't remember

[poisondeathray]

Looking a bit closer - unless the message board display has problems - I think it's because you didn't specify an output file. Add -o output.mp4 -

[ROBO731]

Looking a bit closer - unless the message board display has problems - I think it's because you didn't specify an output file. Add -o output.mp4 -

Thanks, it worked perfectly once I specified the output. I'll check out the audio once it finishes.

[poisondeathray]

I've never used "--fname-from-tag " , but I don't think the metadata will carry through from the original ac3 file since it's a pipe. It will just name it pipe or stdin or something like that if you left it "-" instead of "-o output.mp4 -" . That last "-" is required for the pipe for qaac, but not nero

[ROBO731]

I see, thanks for all the help.

[ROBO731]

It looks like there's one last kink to work out. I'm using the cbr, but the output is coming out as variable.

"D:\ffmpeg-20130724-git-436616f-win64-shared\bin\ffmpeg.exe" -i "D:\t1.ac3" -acodec pcm_s16be -ac 6 -ar 48000 -f s16be - | "D:\qaac_2.19\x86\qaac.exe" --rate keep --threading --raw --raw-channels 6 --raw-rate 48000 --raw-format S16B --quality 2 --cbr 448 -o EncAudio.m4a -

General
Complete name : D:\Encoded Audio.m4a
Format : MPEG-4
Format profile : Apple audio with iTunes info
Codec ID : M4A
File size : 427 MiB
Duration : 2h 12mn
Overall bit rate mode : Variable
Overall bit rate : 450 Kbps
Encoded date : UTC 2013-07-25 17:11:11
Tagged date : UTC 2013-07-25 17:24:49
Writing application : qaac 2.19, CoreAudioToolbox 7.9.8.3, AAC-LC Encoder, CBR 448kbps, Quality 96

Audio
ID : 1
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 2h 12mn
Bit rate mode : Variable
Bit rate : 448 Kbps
Maximum bit rate : 459 Kbps
Channel(s) : 2 channels
Original Channel count : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 426 MiB (100%)
Encoded date : UTC 2013-07-25 17:11:11
Tagged date : UTC 2013-07-25 17:24:48

[Selur]

"but the output is coming out as variable."
which is probably a mistake of MediaInfo,...

[ROBO731]

I see, I was thinking that might be the case, but I wasn't sure.

[Selur]

MediaInfo isn't really good at reporting if content is cbr/abr/vbr (or cfr/vfr for that matter) since it only analyses a headers (and some bytes) of the file and does some educated guessing.

[ROBO731]

Ah, thanks for clarifying.

[ROBO731]

So the audio I'm encoding here is from a dvd and the bit rate is at 448 kbps, what would be a good value to encode it to. I'm thinking 192 kbps or 128 kbps. Also, would it be better to output to a wav format than to have it optimize for mp4 since I'm ultimately going to be muxing this into an mkv?

[poisondeathray]

So the audio I'm encoding here is from a dvd and the bit rate is at 448 kbps, what would be a good value to encode it to. I'm thinking 192 kbps or 128 kbps. Also, would it be better to output to a wav format than to have it optimize for mp4 since I'm ultimately going to be muxing this into an mkv?

128kbps will be too low for 6ch audio most of the time. If you downmix to 2.0 then 128kbps should be ok. 192kbps for 6ch might be ok, but it's going to depend on the source, and the your perception and tolerance of quality , quality of your setup and equipment

For the 2nd question , it really doesn't matter. You can output raw aac and mux it into mkv later. But it won't matter if it's in mp4 container either

[ROBO731]

Well, it downmixed to 2.0 by default, at least according to mediainfo. If it didn't though, what do you think would be a good rate? Also, I'd rather just output to raw aac in that case.

[poisondeathray]

Well, it downmixed to 2.0 by default, at least according to mediainfo. If it didn't though, what do you think would be a good rate? Also, I'd rather just output to raw aac in that case.

Are you sure ? Probably just another (common) mediainfo error . In case you haven't got the "gist" yet, mediainfo makes lots of mistakes

Check by opening the file in something like audacity or an audio editor

Or even listen to the file

[ROBO731]

Yeah, I knew it makes mistakes, but I wasn't sure about which mistakes or of what kind. Audacity doesn't open aac files. I can't quite tell by hearing it, I guess I'll just try encoding it in a few different ways.

[poisondeathray]

Yeah, I knew it makes mistakes, but I wasn't sure about which mistakes or of what kind.

Lots of different kinds

Audacity doesn't open aac files.

It does with the ffmpeg import/export plugin (all kinds of audio that ffmpeg supports)

http://manual.audacityteam.org/man/FAQ:Installation_and_Plug-Ins#How_do_I_download_and...ort_Library.3F

[ROBO731]

Well, I played the file that I encoded at 192 and it sounded fine, when I opened it with vlc and checked the information it said it was 6 channels. So I guess it still is 6.

[ROBO731]

If the delay info isn't stored in the tags, then is there a way to precisely calculate the amount of delay?

[poisondeathray]

If the delay info isn't stored in the tags, then is there a way to precisely calculate the amount of delay?

examine it in an audio editor e.g. audacity

there is no way to accurately "guess" ahead of time. It's different for different files, different aac profiles and sample rates

then you could go back and forth with the --delay and --no-delay option with qaac mentioned above if it really bothers you

[ROBO731]

Ok, thanks for all the help.

[hello_hello]

NeroAacEnc has the biggest initial audio delay ever ,
therefore it's not recommended for the (occasional) reencode of movie audio tracks.
OTOH, qaac has a --no-delay option, to whom this may interest.

I don't fully understand how it all works, but I use foobar2000 for most of my audio encoding, and when converting to AAC with the Nero encoder and M4A or MP4 as the output container, the delay is saved to the container in some way and/or as a chapter file. If you open the encoded M4A/MP4 with MKVMergeGUI it'll show a chapter file. I usually de-select it, but if you don't, extracting it from the MKV after muxing will give you one with a single chapter, usually at 54ms for me. I assume it's the audio delay.

Anyway, the upshot of all that is MKVMergeGUI accounts for the initial encoder delay, so if you remux to MKV it becomes irrelevant (I've no idea what MP4 muxers do). As a result, MKVMergeGUI trims the beginning of the audio stream accordingly and/or sets an audio delay to compensate for what can't be trimmed. According to MediaInfo, after converting audio to AAC and remuxing to MKV, most of my audio streams have an audio delay of 9ms relative to the video. Not enough to worry about even if the player ignores it.

The same applies when I convert with NeroAAC and MeGUI, but I've no idea whether all of the above applies to AAC encoding in general or if it's something specific to the Nero AAC encoder etc. Audio players also seem to be aware of the encoder "padding". Well foobar2000 seems to be, at least. If I open a wave file with foobar2000 and make note of the number of samples, then convert it to AAC, foobar2000 shows the exact same number of samples for the encoded AAC, so I guess it knows not to include the "padding". When converting to MP3 the same applies, although as far as I know MKVMergeGUI doesn't compensate for MP3 padding as it does for AAC.

ROBO731,
Why not use the quality setting rather than specifying a bitrate etc? I guess the principle is the same as x264 CRF encoding, or using one of the LAME VBR presets for MP3 encoding. You specify the quality and the encoder gives you that quality and the average bitrate varies accordingly. I use the default (I think) quality setting for Nero which is -q 0.5 It seems to produce a similar bitrate to LAME's V2 preset for stereo audio and should be at least as transparent. I just tried a couple of quick test encodes while downmixing a 30 minute dts file to stereo. The LAME V2 preset gave me a 27.8MB mp3, while q0.5 for Nero gave me a 28.4MB m4a.

If you're not committed to the command line, one of the benefits of using foobar2000 for converting is you can load a bunch of files into a playlist, right click and select a saved conversion preset, and it'll convert as many simultaneously as you have CPU cores until they're all done.
I haven't played with it much yet, but TAudioConverter can convert with NeroAAC and it's multithreaded.

[ROBO731]

woops, wrong thread.

[kotuwa]

hey, how to swap Left and Right Channels when encoding AC-3 to AAC using Nero?
what are the switches?
Can it be done with BeSweet?
Is there any way or any tool that can do this in AAC tracks without reencoding?

[hello_hello]

I don't think the Nero encoder has switches for swapping channels around.
If you use foobar2000 for re-encoding, you can use the Matrix Mixer when converting (it's a foobar2000 DSP). I only discovered it myself yesterday (I've used foobar2000 for years).
https://forum.videohelp.com/threads/367078-Need-advice-about-mixing-5-1-down-to-stereo?...=1#post2345508
It's configured for downmixing to stereo in the screenshot in the post I linked to, but it's easy enough to use it to swap channels around. By default the input and output channels are all matched. To swap the left and right channels you'd do this:



If you can decode via DirectShow when re-encoding, ffdshow's audio decoder has a filter for swapping channels, or it's mixer filter can do it also.

I don't know what BeSweet can do. If there's a way to swap channels without re-encoding I'm not aware of one, but that doesn't mean it's not possible.