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Old 7th January 2004, 22:46   #1  |  Link
KpeX
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[UPDATED] Audio FAQ

[tebasuna51 edit]
First of all thanks to original owner of this thread KpeX (Last Activity: 4th March 2008 04:19), I want preserve it but update many things to make this FAQ usefull.
There are many links to pages or downloads broken, I want put them with ().
My edits go always between [], when finish my job I'll rename the thread like UPDATED.
[edit end]

A/V Audio FAQ

Welcome to the reorganization of the Audio FAQs. These FAQs address many commonly asked questions. Your first resource in converting should be to follow the Doom9 guides or the more specific audio guides[I can't recommend now these old guides]. Before posting in the audio forum, make sure you've read over any relevant guides and the FAQ in question. There are several people that worked very hard to make these FAQs an excellent source of information, see the last post in this thread for their credit.

The goal of these FAQs is to inform users about how to use the most common audio formats in the AV world and how to encode, decode, and play back their audio with the highest quality, most elegant, and most efficient procedures. Feel free to PM me or post to this thread regarding any corrections or additions.

If you're new to the Doom9 forums, it is strongly recommended to read these threads about our community's netiquette:
Message about Free Pies
Caps and Special Characters
Why a Good Title is Important

As always, search before posting and remember the forum rules, breaking them will be a quick way to get your thread closed or deleted. Cheers and good luck with your audio.

General Audio FAQ

BeSweet FAQ

SVCD/MP2 Audio FAQ

MP3 FAQ

OGG Vorbis FAQ

AC3 & SPDIF FAQ

DTS FAQ

AAC FAQ

[Recommended software]

When was write this FAQ many solutions was based in BeSweet but the last version (1.5b31) was from 2005-9-8 and the last DSPguru post was in 2005-10-7.
It is time to update the soft talking about solutions born in this forum and other free software developpers.
BTW BeSplit can be still used for some operations. Download the last version here: BeSplit v09b8

Dimzon was make plugins for BeSweet, but also create (2005-12-27) a new tool, BeHappy, using AviSynth decoders and filters.
There are a big thread to explain it. Still can be usefull because is easy to update the GUI with the AviSynth tools updated.

After that (2007-5-18) madshi create eac3to, the big thread of this sub-forum (more than 15,000 replies and 10,000,000 views).
Can extract and manage audio streams from multimedia containers VOB/EVO/(M2)TS and MKV.
Like BeSweet is a command line tool but there are some GUI's related in the first post, and also UsEac3to.
But the last version v3.34 was from 2017-11-17 and we can't hope new improvements (read carefull the first post for little updates).

I can't forget Foobar2000, recommended audio player and also with many plugins to manage and encode audio.
It is supported in https://hydrogenaud.io/ fully recommended for audio discussions and tests.

And free open source audio editor Audacity, needed for hard work.
Now the DirectShow filters recommended are the LAV Filters and they are included in a recommended player MPC-HC, (of course there are others VLC, MPC-BE...)
To obtain info from multimedia files we need MediaInfo.
And the recommended container for multimedia files is Matroska.
To rip DVD's (Video or Audio) use MakeMKV, and also BD's.

But the tool than join many multimedia free and open source software is ffmpeg.
Support many containers, AV formats and filters. We can do many things with it.
Here we can talk only for audio management, we can use a GUI like FFMPEG Audio Encoder or also UsEac3to, but I want explain here a little intro to CLI ffmpeg usage.
Maybe can help to the new users to understand part of the power of this tool. The command line syntax is:

Quote:
ffmpeg [global_options] {[input_file_options] -i input} ... {[output_file_options] output} ...
Where:

* [global_options] can be generic options like (only a few samples):

Code:
-v loglevel     set logging level info
-y              overwrite output files
-n              never overwrite output files
-stats          print progress report during encoding
-hide_banner    do not show ffmpeg version and libs included
* The {}... means than accept many inputs, maybe to mux, or mix, or merge, or ...
and produce many outputs, maybe to demux, or split, or ...

* [input_file_options] are applied to each input, only a usefull sample:

Code:
-drc_scale 0        the AC3/DTS decoder must ignore DRC metadata to output the original volume
* [output_file_options] for each output file. Can be a complex syntax, compossed by 4 parts:

[source to be decoded] [filters over uncompresed audio] [codec to encode] [encoder parameters]

All parts are optionals, without info ffmpeg apply the defaults, for instance:

Quote:
ffmpeg -i input.mkv output.mp3
Select and decode the first audio track of the mkv (for instance a DTS-MA-7.1), downmix the 7.1 to 2.0 and encode with libmp3lame at ABR 128 Kb/s.
But if we want decode the second audio track...

* [source to be decoded] that implies know the track order of input files, but most the times the first track [0] is the video track and after go the audio tracks

Code:
-map i:t        where 'i' is the input file order (0 first) and 't' is the track order inside the file (0 first)
-bsf X_core     in audio we can extract only the core from X = dca/eac3/truehd
Quote:
ffmpeg -i input.mkv -map 0:2 output.mp3
Decode the second audio track. But maybe we want override the default downmix method, or resample the audio to create a audio CD, or ...
Now we need apply filters to select our choices:

* [filters over uncompresed audio] can be simple or very complex, for instance:

Code:
-ar rate            set audio sampling rate (in Hz)
-ac channels        set number of audio channels (apply default downmix method)
-vol volume         change audio volume (256=normal)
-af filters         set audio filters, see examples after.
-filter_complex fc  when a complex syntax is required. Examples after.
Simple audio filters examples:
Code:
-af "aformat=sample_fmts=s16"           Downsample bitdepth
-af "adelay=delays=1600:all=1"          Delay 1600 miliseconds inserting silence
-af "atrim=1.1:8.2"                     Trim audio between 1.1 and 8.2 seconds
-af "atempo=0.959041"                   Slowdown audio (25->23.976) preserving the pitch
-af "aresample=50050, asetrate=48000"   Slowdown audio (25->23.976) changing the pitch
-af "pan=stereo|FL=.3254c0+.2301c2+.2818c4+.1627c5|FR=.3254c1+.2301c2-.1627c4-.2818c5"  dplII downmix
Like you can see we can concatenate two simple filters with a comma ','

Complex audio filters examples:
Quote:
-filter_complex "asplit [f][s]; [f] atrim=0:11 [ff]; [s] atrim=11, adelay=delays=2000:all=1 [ss]; [ff][ss] concat=v=0:a=1 [a]" -map "[a]"
Insert 2 second of silence after the first 11 seconds

-filter_complex "asplit [f][s]; [f] pan=3.1|c0=c0|c1=c1|c2=c2|c3=c3 [r]; [s] pan=stereo|c0=0.5*c4+0.5*c6|c1=0.5*c5+0.5*c7, compand=attacks=0:decays=0oints=-90/-84|-10/-4|-6/-2|-0/-0.3, aformat=channel_layouts=stereo [d]; [r][d] amerge [a]" -map "[a]"
Downmix 7.1 -> 5.1 presenving the volume, and quality, of first channels and downmix only the surround channels trying to preserve the balance most the time between front and surround channels.
I want explain the more complex last one by parts (see also):
Code:
"asplit [f][s];                                        I use the input two times named now [f] and [s]
[f] pan=3.1|c0=c0|c1=c1|c2=c2|c3=c3 [r];    The four first channels remain untouched and named now [r]
[s] pan=stereo|c0=0.5*c4+0.5*c6|c1=0.5*c5+0.5*c7,  The four last channels are downmixed at half volume
compand=attacks=0:decays=0:points=-90/-84|-10/-4|-6/-2|-0/-0.3,  And are amplified by 2 at low volumes
aformat=channel_layouts=stereo [d];                  The output are formated like stereo and named [d]
[r][d] amerge [a]" Now merge the first 4 chan [r] with the 4 surround downmixed to 2 [d] and named [a]
-map "[a]"                                                           We select [a] to be encoded after
Now the semicolon ';' implies changes in named input and output

After manage the uncompressed samples we want select the output format or codec

* [codec to encode] To know all see the docs.

Code:
-strict -2      Some experimental encoders need this parameter: mlp, truehd, dts
-acodec codec   And: aac, ac3, alac, copy(to extract), eac3, flac, mp2, mp3, opus, pcm_s24le, ...
-f format       For instance to specify the header of a pcm_s24le, we can select wav or w64
* [encoder parameters] See the docs to know all parameters allowed

Code:
-aq quality             set audio quality (VBR codec-specific)
-ab bitrate             set audio bitrate (CBR or ABR)
...
-center_mixlev 0.707    codec specific example for ac3 (recommended)
But if you need a specific encoder or parameter, remember you always can use the 'pipe' method, for instance:

Code:
ffmpeg -i INPUT -map 0:T  FILTERS -acodec pcm_s24le -f wav - | qaac --ignorelength --adts --no-delay -V 91 -o OUTPUT.aac -
Remember to select the codec and format supported by the STDIN of encoder.

I hope this post work like a intro to ffmpeg, in the rest of FAQ I add other examples of use it.

Last edited by tebasuna51; 30th September 2021 at 11:35. Reason: update
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Old 7th January 2004, 22:46   #2  |  Link
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General Audio Procedures FAQ
Cross-format techniques for encoding, decoding, and playback.

1. What is the difference between lossy, lossless, and uncompressed audio formats?

Uncompressed audio is simply audio without any compression applied to it. Uncompressed audio is commonly used in AV conversions in PCM or WAV form.
[WAV is a container than support audio data, not only PCM uncompressed samples]

Lossless audio applies a compression to uncompressed audio without losing any information or degrading the quality at all. Lossless audio is not common in the AV world, but it is possible with formats like WMA Lossless or FLAC in Matroska.
[Also TrueHD and DTS-MA are lossless and used now in BluRay Disks]

Lossy audio attempts to apply to discard as much 'irrelevant' data as possible from the original, with the goal being to produce a file much smaller than the original that sounds almost identical. This results in a much lower bitrate and filesize then lossless or uncompressed audio. Lossy audio formats are extremely prevalent in AV, and include AC3, DTS, AAC, MPEG-1/2/3, Vorbis, and Real Audio.
[Now Vorbis is replaced by Opus]

We also talk about lossy and lossless processes. Whenever you transcode to a lossy format (for example wav > MP3), a loss in quality follows, therefore it is a lossy process. Transcoding from a lossy format to a lossy format (for example Mp3>AAC) is even worse, since there is loss introduced both from the first lossy file and the encoding of the second.

2. What's CBR/ABR/VBR?

CBR means that the stream's bitrate is constant and never changes.
VBR means that the stream's bitrate is variable, and changes in according to the amount of information that needed to be encoded.
ABR has a variable bitrate for each frame, but its average bitrate is a constant.

Conclusion:
CBR is a private-case of ABR which is a private-case of VBR.

[For audio movie tracks the VBR mode is highly recommended, because silences and dialogs only need low bitrate and we can reserve bitrate size for parts than need it.]

3. What's the difference between stereo, joint stereo, and dual channel?

Dual channel has 2 mono-channels, meaning each channel is encoded with half the bitrate of the overall bitrate.
Stereo has 2 seperate channels, but the bitrate allocation between those two channels changes accordingly to the amount of information there is in each channel.
Joint stereo has 2 channels, but takes advantage of what is common between the channels, so the compression gain is higher.

4. What are the different kinds of joint stereo?

Two of the most used joint stereo modes are IS (intensity stereo) and M/S (Mid/Side). M/S matrixing computes the sum and the difference of the two audio channels and stores these as two channels. This method is very efficient and is a lossless process, which means the two original channels can be extracted exactly as they were.

Intensity stereo replaces the left and right channels with a single signal plus directional information. This method is lossy and destroys DPL information. This method is only recommended at low bitrates.

Many encoders can use a combination of full stereo and either or both of these methods, deciding which to use on a per-frame basis.

5. How can I transcode AC3 5.1 to Pro Logic/Pro Logic II?

There are two steps to having a 2.0 channel stereo file that contains Dolby Pro Logic (II). First, a dolby pro logic source is needed, in this case we apply a DPL downmix on the 5.1 source. This can be done easily with besweet, in the '-azid()' section add a '-s surround' for Pro Logic or '-s surround2' for Pro Logic II.

[Instead Besweet you can use now ffmpeg, eac3to or BeHappy, for ffmpeg or eac3to there are many GUI's, I can support UsEac3to, by CLI I can recommend ffmpeg (replace uppercases with desired values):

ffmpeg -drc_scale 0 -i INPUT51 -af "pan=stereo|FL=.3254c0+.2301c2+.2818c4+.1627c5|FR=.3254c1+.2301c2-.1627c4-.2818c5" -acodec AC3 --dsur_mode 2 -ab 192k OUTPUT.AC3
]

Secondly, when the audio is encoded, the joint stereo information must be preserved. In order for this to happen, either full stereo or M/S joint stereo must be used - Intensity stereo will destroy DPL information. More information on Dolby ProLogic & Dolby Surround can be found here.This can be done with all the common audio formats:
  • MP3: Use the lame encoder, and use joint stereo. See the MP3 FAQ #9.
  • Vorbis: Use lossless or light lossy channel coupling.
  • AAC: Use M/S joint stereo, which pretty much all the AAC encoders use AFAIK.
  • MP2: MP2 specs do not support M/S, so you'll have to use full stereo.
6. Where can I find more information about audio coding formats and techniques?

http://www.hydrogenaudio.org [Recommended for audio soft and test, and foobar2000 player and converter forum]

7. What is the best lossy audio format for me to use?

As you should know, forum rules prohibit asking what's best. In general your own ears are the best answer to this question.
[Test in hydrogenaudio, test including Opus, multichannel test

To select a lossy encoder the first parameter is the compatibility with the device player you want use.
For a PC or a power standalone player conected by HDMI to a modern AVR all codecs can work, but others can have troubles.
Now the AAC is the more compatible with stereo, and AC3 for multichannel.

After that and checking the listening test the second parameter can be the ratio quality/size and the order can be: Opus>AAC>EAC3>AC3.
Also remember than there are differences in the listening test at low bitrates but the differences at high bitrates are low.]

8. When converting, is it better to downsample my DVD audio to 44.1 KHz or keep it at 48 KHz?

Unless you have to meet a standard's specifications, such as 44.1 KHz for SVCD, there is usually no reason to downsample audio to 44.1 KHz. It is possible that downsampling will introduce rounding errors when finding new sample points, which will degrade the quality, but when using a high quality resampler such as SSRC, this quality loss will be extremely miniscule. It can also be argued that certain audio encoders, such as Lame, are better tuned for 44.1 KHz, but these differences are also very miniscule.

9. How can I change the samplerate of a wave file?

Download SSRC from here, and use the following command:
Code:
ssrc --rate 44100 input.wav output.wav
In this case, the destination sample rate is 44.1 Khz (44100 Hz).
[We can use now ffmpeg, eac3to, BeHappy (supported in this subforum) and others. The CLI command:
ffmpeg -i INPUT.wav -af "aresample=44100" OUTPUT.wav]

10. What is SPDIF [/HDMI] and when is it used?

SPDIF (Sony Phillips Digital Interface) is a physical digital interface (_not_ analog). The connection can be either coax or optical (fibre) meaning in general it should provide less noise interference from my experience (varies depending on hardware and source). The SPDIF interface can pass PCM [only 2.0], DTS or Dolby Digital streams to your receiver/amplifier for decoding. In theory, your SPDIF capable sound card passes the audio packets without modification (It’s theory because much discussion has taken place as to if Creative's sound cards alter packets).

[HDMI is the new physical digital interface, with special cable, to transfer Audio/Video (AV) between devices.
Talking of audio, support the same than SPDIF plus TrueHD, DTS-HD and PCM multichannel]

11. Is it possible to backup music from a concert dvd into an audio cd?

Yes, the AC3/DTS needs to be demuxed and then converted to a PCM-WAV, Dolby Digital-Wav, or DTS-Wav.

Checkout BeSure from doom9's download page. It can create DTS-CD, AC3-CD & plain CDDA.
[Instead BeSure, BeSweet (with old ac3enc) and Surcode (commercial DTS encoder) you can use now ffmpeg to resample and encode to AC3 or DTS:
ffmpeg -i ANY_INPUT51 -vn -af "aresample=44100" -strict -2 -c:a dca -b:a 1411.2k OUTPUT.dts
or
ffmpeg -i ANY_INPUT51 -vn -af "aresample=44100" -c:a ac3 -b:a 640k OUTPUT.ac3
and after convert to Ac3-Wav or DTS-Wav with spdifer, a tool from AC3 Filter Tools
spdifer OUTPUT.ac3 OUTPUT.wav -wav
]

12. When transcoding 5.1 AC3 to 2-channel audio, which AC3 stream should i decode? 5.1 or 2.0?

The 2.0 stream has a better mix for 2 channels than the 5.1 to 2.0 downmix features offered by AC3 decoding tools. Therefore, it should be best to work with the 2.0 stream. Transcoding the 5.1 stream and downmixing it manually will only be useful for users who have a surround system that has a Dolby ProLogic (II) decoder. Some 2.0 tracks aren't downmixed for Pro Logic, so you must take the origianl 5.1 track and downmix it with a transcoder that is capable of doing it. More info on Dolby ProLogic & Dolby Surround can be found here.

13. How can I extract an audio stream (AC3/MP3/OGG/WAV) from an AVI file?

Download VirtualDubmod and open your AVI file. Click streams -> stream list. Click on the audio stream you want to extract; then click demux to remove the stream losslessly, or click decode to decompress the audio to a .wav file. This will work for any audio stream in an .avi file (first/second etc.)

[We can extract an audio stream X (order) from any container with:
ffmpeg -i ANY_CONTAINER -map 0:X -acodec copy OUTPUT.EXT]

14. How can I split my AC3/DTS/AAC/MP3/MP2/MPA/WAV track into several seperate, shorter, tracks?

Use BeSplit, it was written especially for that. Note that (Chapter-X-tractor) can generate BeSplit's commandline for .ifo files.

An alternative for DTS is (DTS Trimmer/Concatenator).

[The last BeSplit version (v09b8) can be download here.
Other tool to split some audio files is DelayCut]

15. My AVI plays with no audio. What can I do?

First of all, reread the forum rules, you will get no help with downloaded or illegally obtained files here.

If the file was created by yourself, you'll simply need to install a DirectShow filter capable of rendering the the audio format. Check Doom9's download page for some common audio filters. GSpot, also available on the download page, can be of assistance in identifying what audio is contained within an AVI.

[Now the DirectShow filters recommended are the LAV Filters and they are included in a recommended player MPC-HC, (of course there are others VLC, MPC-BE...
To obtain info from multimedia files we need MediaInfo]

16. Please post a list of the most recommended winamp plugins.

[Like Audio player I recommend now Foobar2000]

17. Can I decrypt/rip/create a DVD-A (DVD-Audio) disk? What format does DVD-A use?

[Use MakeMKV to decrypt/rip and play the mkv's without problem]

DVD-A uses either LPCM or a lossless encoding called MLP (Meridian Lossless Packing), and supports up to 24 bit and 192 KHz audio.

As far as creation goes, the dvd-audio project aims to provide audio enthusiasts with a set of free software tools to enable the authoring of DVD-Audio disks compliant with hardware DVD-Audio players, plus a software player for such disks. Note that this project is still in an early stage of development.

If you have deep pockets, minnetonka's (discwelder chrome) is a commercial DVD-Audio authoring application.

18. Is it possible to use Musepack (MPC) audio with video?

At this time it is not yet possible in any container format. The developers of Matroska have plans to include Musepack audio in Matroska, but due to the opinion of the lead developer of Musepack and the bitstream format of Musepack this will not happen until the SV8 bitstream (the next version of Musepack) is finalized. However there may be a Musepack SV7>'SV7.5' bitstream converter, so your current MPC files might be possible in Matroska in the future.
[Musepack SV8 still without support]

19. How can I convert the frame rate of my audio with freeware?

First of all, for the nth time, audio does not have a framerate. When someone talks about changing the framerate of audio, they are talking about time stretching the audio to match a video framerate that has been converted. FRC conversions can be done with several commercial tools, but that is not the focus of this FAQ, use google to find these.

A 'framerate conversion', which we will refer to as a speed changing operation, can be done either with or without pitch correction. When done without pitch correction, if the speed change is large, you will notice that the audio sounds too high or low pitched, much like changing the speed of a tape or record player. Such an operation can be done with BeSweet, simply add a
Code:
-ota( -r 23976 25000 )
to your ota section of your besweet commandline, where in this case the original video framerate was 23.976 and the final framerate is 25.000.

A speed change with pitch correction is also known as a tempo change; the speed at which the audio played back changes without chanigng the pitch from the original. to do this, you can use the freeware tool Audacity, or use BeSweet v1.5b28 or newer with the (commandline) :
Code:
-soundtouch( -r 23976 25000 )
[ffmpeg can also do the job with a filter, speedup without change the pitch: -af "atempo=1.042708"
speedup changing the pitch: -af "asetrate=50050, aresample=48000"
There are also SoundStretch]

Last edited by tebasuna51; 30th September 2021 at 13:23. Reason: update
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Old 7th January 2004, 22:47   #3  |  Link
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SVCD/MP2 Audio FAQ
MPEG-1 Layer 2 & MPEG-2 Multichannel Questions

1. What formats are supported by SVCD specification?

SVCD supports two audio formats officially:

-CBR MPEG-1 Layer 2 Audio, 32-384 kbit/s, 44100 Hz sampling rate, 2 channel, up to 2 tracks

-MPEG-2 Multichannel Audio (although allowed in the spec, it is supported by very few players)

2. Is it possible to have 5.1 channel audio on my SVCD?

Yes, there are two ways. The most reliable is to use a Dolby Pro Logic or DPLII downmix when encoding to 2-channel MP2. Simply add a -surround or -surround2 to your besweet commandline in the azid section.

The other way is to create a MPEG-2 Multichannel 5.1 stream. Standalones that can decode these streams, however, are few and far between.

3. How can i convert AC3 into Mpeg2multichannel?

You would need the phillips (encoder), and BeSweet. Here are the two commandlines :
Code:
BeSweet -core( -input input.ac3 -output inter.aif -6chaiff ) -ssrc( --rate 44100 ) -ota( -g max ) 
pub_enc -l 2 -m s -n d -r 44.1 -b 384 -L -p 3 -e -v 0 inter.aif final
The resultant file will be final.mpg which is a mpeg2multichannel stream.

[If anyone have the phillips encoder maybe we can put a new link, BTW I think than nobody need that old surround method not fully supported]

4. What MP2 encoders are available?

The two most common encoders (both included with besweet) are TooLame and MP2Enc. Both are open source, and MP2enc is the encoder from the CDex project. Although extensive testing has not been done, both provide good and similar quality.

[Of course ffmpeg can encode to .mp2]

5. Is it possible to increase the volume or normalize MP2 without decoding & reencoding (i.e. losslessly)?

Yes, the only tool that offers this feature currently is BeSplit. An example command line is
Code:
BeSplit -core( -input infile.mp2 -output outfile.mp2 -type mp2 ) -ota( -G max )
6. Which psymodel should I use with toolame or MP2enc?

There is much debate about both the psymodels, and the only fully correct answer is to use your own ears and judge for yourself.

In both toolame and mp2enc, Psychoacoustic Models 1 and 2 are the ISO standard models. Although they are standard, they are not to be considered very high quality. However the psymodel does not have a large impact on the sound quality. In general, Psymodel 2 is considered to be of better quality, but may have problems with frequency cutoff and tonality detection. A guideline used by many users is to use psymodel 2 at bitrates less than or equal to 160 and psymodel 1 at higher bitrates.

Later versions of toolame have implemented several new psychoacoustic models: psy 3 is a re-implementation of the standard psy 1 and psy 4 is a cleaned up version of psy 2. Although these versions will not sound much different from their ISO standard counterparts at the moment, there is possibility of their improvement in the future. See this page for more information on toolame's psymodels.

7. Should I use stereo or joint stereo for MP2?

In general it is recommended to use stereo for MP2, the joint stereo in MP2 is intensity stereo only, not M/S (mid/side joint stereo as used in lame and many other codecs), so joint stereo is probably a bad idea except at very low bitrates (less than 128kbps).

8. I have a dual channel MP2 with a different language in each channel. How can I separate the languages?

BeSplit v0.9b6 and up can accomplish this losslessly by producing two mono .mp2 streams. Note that this is only possible if your mp2 is dual channel. Use besplit from commandline:
Code:
BeSplit -core( -input stereo.mp2 -prefix e:\channel -type mp2 -demux )

Last edited by tebasuna51; 25th September 2021 at 18:54. Reason: update
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Old 7th January 2004, 22:48   #4  |  Link
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MP3 FAQ
MPEG-1 Layer 3 encoding and playback

1. What is MP3?

MP3, also known is MPEG-1 Layer 3, is a general purpose lossy encoded audio format capable of high compression rates retaining considerable quality. Layer 3 is not a new version of MPEG Layer 1 or Layer 2, but a different complexity level. There are many MP3 encoders available, the highest quality encoder is Lame, which is also free and open-source.

The Lame encoder version 3.9x is highly recommended over any other MP3 encoder.

2. Does MP3 support more than 2 channel audio?

The MP3 specification does technically support multichannel, but implementations are rare and compatibility is unsure. Keep in mind that you can encode your multichannel source with a pro logic (II) downmix into stereo MP3. Fraunhofer has made (a 5.1 MP3 encoder/decoder demo available), although this is not a true 5.1 implementation but rather a backwards-compatible matrixed format.

[The link is broken, and the format is deprecated]

3. Can the volume of an MP3 be raised up without reencoding?

Yes, there are few programs that offer this, check mp3gain & mptrim for instance.
BeSplit offers this as well. for commandline, read q5 in the MP2 FAQ.

4. How can I achieve good quality results with MP3?

For movie soundtracks 128kbps and above will usually have acceptable quality. To increase quality, use VBR and joint stereo. The recommended commandline for transparent quality with the Lame encoder is -V2 (which can be used in BeSweet (-V settings require lame 3.94 or later). The recommended version is lame 3.97. This preset will result in transparency with the vast majority of samples, and usually a bitrate of 140-170 kbps for movie soundtracks.

[Of course the versions are updated until 3.100, and ffmpeg are updated also (now Lavc59.9.100 libmp3lame), and can encode mp3 with -aq 2, equivalent to -V2 for Lame]

5. What version of Lame is included with BeSweet?

The latest versions of BeSweet now include Lame 3.90.3, the previously recommended version. Recent full packages of belight include lame 3.97; this version is highly recommended for quality, see Q8 for more information. As mentioned before, any lame encoder version 3.9x is a good quality choice. Most lame users are using the latest stable version or 3.97.

[Now the last version is 3.100]

6. My VBR MP3 file shows the wrong length. How can I fix this?

If you are determining the playback length from a DShow based player such as WMP or ZoomPlayer, the wrong length is because of the buggy Microsoft MP3 Dshow filter that comes with windows. Ignore it or use a good non-dshow player such as winamp.

If the length still appears to be wrong, use BeSplit, with the following command :
Code:
BeSplit -core( -input c:\track.mp3 -prefix c:\track -type mp3 -fix )
This solution is also good for wrong-reported-length wav files. just switch "-type mp3" with "-type wav".

[The players can't know the correct duration of VBR files if not exist a special header field than inform of that.
If the mp3 was make VBR but without that info it is not possible know the full lengh without read the full file.
The same with AAC VBR files, we need put in a m4a container to insert that info]

7. Why is my MP3 destination file downsampled to 32kHz or lower?

In latter versions of LAME(3.9x), it will automatically downsample audio for low bitrates(<128kbps) to reduce artifacts.

8. Where can I find lists of recommended MP3 settings?

List of recommended LAME settings

9. Will encoding to MP3 with joint stereo preserve Pro Logic/Pro Logic II information?

Lame, the only recommended MP3 encoder, uses a high quality M/S (mid/side) joint stereo mode, which is much better than the intensity stereo used by the joint stereo mode of many other encoders like MP2. Due to the way that DPL surround channels are matrixed into the two stereo channels, this M/S joint stereo mode is actually recommended when using Pro Logic. Additionally, the --alt-presets are highly recommended for DPL. See a more detailed explanation here.

10. How can I properly mux a VBR MP3 into AVI?

Use VirtualDubMod or Avi-Mux GUI.

[Of course ffmpeg can do the job also, but formerly the avi container not support audio VBR and is always recommended mux in mkv container:
ffmpeg -i input.avi -i vbr.mp3 output.mkv
Or use MkvToolNix]

11. What about encoding MP3 under Linux/*nix?

Lame, the most recommended MP3 encoder, builds under GNU/Linux and a host of other platforms. Grab the source from sourceforge, or get rpms from rpm.pbone.net or debian builds from (www.rarewares.org/debian.html)

Once lame is installed, mencoder, part of the mplayer project, has an mp3 encoding module based on lame. Simply reconfigure and rebuild your mplayer source, and when encoding with mencoder you'll be able to encode to mp3 using the -oac mp3lame options. For more information see the mplayer man page under the 'codec specific encoding options' section.

[I not found the rarewares link, I'm not a linux user but see in https://lame.sourceforge.io/ , and ffmpeg work also in Linux (Debian, Ubuntu or Fedora pakages)]

Last edited by tebasuna51; 26th September 2021 at 01:43.
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Vorbis & OGG FAQ [Replaced by Opus]

1. What is OGG? What about Vorbis? What's the difference?

[Like we can see in wikipedia Opus replace Vorbis (and Speex) and can use OGG container or also MKV, WEBM, MPEG-TS and MP4.
It have better performance and compatibility than Vorbis. See also both encoders in hydrogenaudio wiki: Opus and Oggenc]

Vorbis is an open-source general purpose lossy encoded audio compression format developed to be free and completely unpatented.

OGG is the native container for Vorbis, it can also contain other audio formats such as FLAC and Speex. Referring to OGG audio is technically incorrect just as referring to AVI audio or MP4 audio is incorrect, all of these refer to containers. You may have also heard of OGM, which is a modification of OGG to support video, other types of audio, chapters, subtitles, and more. For more information see the OGM FAQ.


2. How can I encode to Vorbis?

[Only the Rarewares links work and have the last recommended Vorbis encoders. Also the Opus tools, and here Opus 1.3.
And ffmpeg support also Vorbis and Opus codecs. The rest of links are oldies.]

You can download OagMachine, BeSweet or HeadAC3he to transcode your AC3 into an Ogg Vorbis file. As usual Doom9 has ()an excellent guide to get you started. A good start point for quality(q) settings is around q2 and this will provide you with a file that averages 96kbps. Unlike MP3, when discussing Vorbis, we dont mention bitrates but discuss in terms of -q modes(quality).

As vorbis is an open-source audio compression format, there are many encoders available, and although they are often referred to as 'forks', they are fully compliant Vorbis encoders. A list of the most common encoders follows:

1) Xiph 1.1.1: The 'official' Ogg Vorbis encoder. (Download link).
2) aoTuV: vorbis tuning by Aoyumi, found to have excellent performance in listening tests. Aoyumi's earlier tunings are now incorporated into the official libvorbis 1.1, but further betas are available from Aoyumi's site.(Homepage), libvorbis dlls.
3) QKTune: A vorbis tuning by QuantumKnot. (Download).
4) GT3b2: A vorbis tuning by Garf. (Download).

There are many more tunings available, the best source for your Vorbis software needs is usually to check out the Vorbis page at Rarewares.


3. How can I upgrade BeSweet or HeadAC3he to a different version of Vorbis?

For BeSweet: If BeSweet finds libvorbis.dll in the besweet directory, it will use that instead of the OGG Vorbis DirectShow filters. Note that for some compiles you'll also need libmmd.dll in the same directory. Get them both from RareWares.

For HeadAC3he: HeadAC3he uses a special vorbis.dll for encoding, get it once again from RareWares.

[We don't need now BeSweet or HeadAC3he]

4. GT3b1? 1.0.1? Which encoder version should I use? What are the best settings?

[Hydrogenaudio wiki for Vorbis and Opus]

Different encoder versions have different bitrate tunings. Vorbis 1.x is well tuned for low bitrates, GT3b1 has some high-bitrate enhancements, and development continues of course. AoTuV versions perform consistently superior in vorbis listening tests. Combined versions are available as well. In general, the -q settings are excellent VBR presets and usually don't need additional parameters. For more information see (Hydrogenaudio's recommended Vorbis encoders and settings).

5. Is it possible to transcode to 5.1 channel vorbis?

Yes, Vorbis supports multichannel audio. Correct channel coupling has yet to be implemented as of Vorbis 1.1.1 so in general it is not recommended when transcoding from AC3; the bitrate required to get a good quality 5.1 Vorbis stream will likely be close to as high as the original AC3 stream.

[Of course Opus also support multichannel]

6. What containers will allow me to use Vorbis + Video? Will Vorbis in AVI ever work?

Currently, Matroska and (OGM) support Vorbis with video quite well. Vorbis in AVI will probably never work due to the complexity of Vorbis streams and the difficulty of correctly synching such a stream in AVI. However, AVI works with many other things that it's not supposed to so who knows.

[Opus is supported in MKV, WEBM, MPEG-TS and MP4]

7. Is it possible to pass 5.1 vorbis over S/PDIF?

There are not any known receivers or amps that support Vorbis decoding, so raw Vorbis cannot be passed over S/PDIF. However there are several filters that support reencoding six channel audio to AC3 on the fly: (Dolby Out DS) filter by Haaan , and AC3Filter can be used to encode AC3 over S/PDIF on the fly. Although this reencoding process will theoretically result in a quality loss, using the full 640 kb/s bitrate allowed will result in an excellent listening experience.

[The same for Opus, but now with HDMI (instead SPDIF) conexion the players can send PCM multichannel decoded from Vorbis or Opus]

8. How can I encode to OGG vorbis under linux / *nix?

The vorbis tools ( for encoding and decoding ogg ) are part of most modern linux distributions, check your distributions' repositories for binary packages. As always debian users can get plenty of vorbis apps from (www.rarewares.org/debian.html.) You can also grab and build the source from vorbis.com. For basic encoding/decoding needs, you'll want to install libogg, libvorbis, and vorbis-tools (in that order).

Vorbis encoding is also supported in (VLC) and several other encoding tools.

[And also with ffmpeg for linux]

Last edited by tebasuna51; 29th September 2021 at 10:11.
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Dolby Digital/AC3 FAQ [And DD+/EAC3]


General AC3 Procedures Questions

1. What is AC3? Dolby Digital?

AC3, Dolby Digital, Dolby Surround AC3, A/52, and Audio Coding version 3 are all names for the same general purpose lossy encoded audio format developed by Dolby Laboratories. AC3 is most commonly used in 5.1 channel form and is the successor of Dolby Pro Logic, the difference being that AC3 carries 6 discrete channels whereas Pro Logic had matrixed surround channels. AC3 has great popularity, and is used in LaserDisc and DVD formats at 384-448 kbps and also quite commonly in cinema, usually at 640 kbps.

[Dolby Digital Plus is a improvement for more quality, bitrate, channels, programs and also the Atmos system to obtain surround 3D with specifics AVR's.]

2. How can I transcode/encode AC3? Are there any freeware encoders?

Sonic Foundry Soft Encode, one of the most popular AC3 encoders, is no longer developed, the AC3 encoder is now available as a plugin for Sonic Foundry's Acid.

Surcode also sells several good AC3 and DTS encoding softwares.

Of the several commercial encoders available, all known commercial encoders are based on the Dolby software encoder. SonicFoundry SoftEncode and other older AC3 encoders use Dolby v 6.6.2, newer encoders such as the SonicFoundry Acid plugin use Dolby v 7.

The only free encoder is part of the ffmpeg project and is generally considered to be not as high quality as a commercial encoder using the Dolby libraries. However, incompatibilities with AC3 specs should be solved now, and using BeSweet 1.5b26 or later is a good choice for free, multichannel AC3 encoding.

It’s not recommended to transcode a 5.1 AC3 file below 224kbps due to quality degradation.

[Now ffmpeg AC3 encoder have enough quality to be used without troubles, but like EAC3 have still some limitations, of course without Atmos improvement]

2b. What about AC3 encoding under linux/*nix?

FFmpeg, a native linux application, supports AC3 encoding via libavcodec, as does mplayer, vlc, and other libavcodec-based apps. An AC3 transcoding example commandline is:
Code:
ffmpeg -i input.ac3 -acodec ac3 -ac 6 -ab 256 output.ac3
3. What's the difference between normalization and dynamic range compression?

Normalization applies gain to all the audio, essentially multiplying everything by a common factor in order to max out the use of the 16 bits you have in most wav files. Normalization preserves the dynamics of the original recording, meaning that the relationship between loud and soft stays the same, except that everything is now louder by the same factor.

Dynamic range compression basically amplifies softer parts more than louder parts, to bring soft and loud closer together and lessen the contrast between, say, explosions and background music.

[Both concepts, Dialog Normalization (DN) and Dynamic Range Compression (DRC) was introduced by Dolby with good intentions but with bad results, because other sources go to inverse direction (see Loudness war).
To compete in volume with other sources I recommend supress the DN attenuation (like eac3to and other tools do) included in the AC3 metadata.
And ignore the DRC metadata info (than some encoders include, not ffmpeg) to avoid lose quality when decode: use always -dsc_scale 0
The moderns AVR's have enough capacity to apply 'Night mode', or equivalent, on the fly when it is necesary]

4. Where can I get some comprehensive information on encoding AC3?

There is an excellent guide available courtesy of SomeJoe.

[Like I say before I can't recommend to be Dolby compliant because other sources aren't (Audio CD's, commercials in TV), the best option is encode without DN and DRC like other codecs do (DTS, MP3, AAC, Opus, FLAC,...)]

5. How can i join AC3 files?

Let's say you want to join stream1.ac3,stream2.ac3,stream3.ac3 into stream.ac3.
create a listfile. join.lst:
stream1.ac3
stream2.ac3
stream3.ac3

Use BeSplit :
Code:
BeSplit -core( -input join.lst -output stream.ac3 -type ac3 -join )
[AC3, like other streams with a header in each frame (DTS, AAC) can be joined with a simple binary copy:
copy /B stream1.ac3 + stream2.ac3 + stream3.ac3 full_output.ac3
Of course the output is only valid if the parts have the same Bitrate, Samplerate and Channels]

6. What can I do if I'm having trouble adding an AC3 file to a DVD project?

First try fixing it with BeSplit. If there is still a problem, it could be a VBR AC3 that your authoring program doesn't support so reencode it to CBR (see Q#2 for AC3 encoders).

[Sometimes an AC3 stream is invalid whith fragments a different bitrate/channels.

I make a tool (LeeAudBi) to detect errors, to work with the attached one you need the autoit3 package installed to interpret the .au3 files.
If you want the compiled exe you need download it from here (to big to be attached).

There are also other tools to split fragments stereo from multichannel typical from AC3 TV capture.

And tools to modify Dialog Normalization (AC3 or EAC3) or set the AC3 flag DPL encoded.

Finally WavSplit (in channels) and WavFix (to repair WAV headers)]
Attached Files
File Type: 7z AC3Util.7z (92.7 KB, 16 views)
File Type: 7z Wav2Util.7z (15.3 KB, 7 views)
File Type: 7z LeeAudBi7i.7z (72.2 KB, 12 views)

Last edited by tebasuna51; 4th October 2021 at 03:48. Reason: update
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AC3 Playback Questions

7. Which software do I need to install for AC3 playback?

Download Valex's AC3filter audio filters and install. This filter is highly recommended over any other AC3 capable filter, it's very feature rich and best of all it's open source. Please read over his excellent documentation for a full understanding of all the capabilities of the AC3filter. Valex frequently updates the package so you might check Doom9's news or the AC3filter page often.

Secondly, you will need to use your favorite media player to playback the media container (ie. OGM, MKV, or AVI) which has the embedded the AC3 track.

[At this moment the AC3 codec is the most compatible with many devices, in the PC you only need a player like MPC-HC or VLC.
Maybe the question is how you have the conexion betwen the PC and your audio equipament, and what support that audio equipament:
- A modern AVR conected by HDMI: TrueHD/DTS-HD/EAC3/AC3/DTS passtrough, other formats decoded to PCM multichannel.
- A old AVR or only conected by SPDIF: DTS/AC3 passtrough, other formats downmixed to PCM 2.0 or recoded on the fly to AC3 with AC3Filter or similar.
- Only a analog stereo equipament: all formats must be decoded and downmixed by the player.

With LAV Filters or a correct player the rest of questions in this post are oldies.]

8. How do I determine if the filters are registered and installed correctly?

If the filter isn't installed correctly you'll typically see an error when playing back the media file such as “DirectShow can not render the file”. Open GraphEdit (can be found in Doom9's download section), select Graph>Insert Filters (Ctrl-F). A dialog should appear and you can double click the Direct Show category. This will display all DS filters that the system has registered and are available for use. Double click on a filter to see the path where the filter file resides on your drive (ie. C:\Program Files\AC3Filter\ac3filter.ax). If the filter doesn't show up in GraphEdit then most likely a problem with the install occurred and you should attempt an uninstall of the AC3filter and then reinstall using the supplied installer.

If the filter still isn't being loaded in your media player or GraphEdit then try manual registering. First verify where the filter physically resides by doing a search for ac3filter.ax, typically it will be in the path I note in the next line. Then open a DOS window prompt->type C:\Program Files\AC3Filter\. At this point we can now register the filter by typing regsvr32 ac3filter.ax, you should see a dialog pop-up indicating the filter registered properly. As an alternative you could try the Moonlight Odio Dekoda from Elecard here (has much less features than Valex's AC3filter but better than Intervideo)

9. Why don’t I hear any sound or the sound is garbled during playback?

Typically this problem is due to the way your system or media player is configured. The quick fix might be to check if your media player is using “DirectSound” or “WaveOut”. Most systems use “DirectSound” by default and will work fine but if you experience problems then try using “WaveOut” instead. Some users have also reported they have to use the Default Direct Sound device selected for proper playback (ie. Default Direct Sound Audigy). To switch DirectSound/WaveOut output, use the WaveOut fix package available on the Doom9 download page.

10. Why does the sound sometimes stop in the middle of my movie and then come back on?

Drop outs of audio over SPDIF can be caused by a poor quality power supply or you may need to purchase a surge protector to regulate current. Also, audio drop outs can be caused by a poor quality optical or coax cable. Some users have reported poor PC power supplies to be the cause as well. As a last thought, I have experienced drop outs if broken muxing tools such as NanDub was used with too tight of settings (ie. 64/64ms). More about muxing the AC3 follows below.

11. The AC3’s volume seems low during playback, what can I do?

Check your receiver manual to see if your amp supports DRC(Dynamic Range Control) which can increase the volume dynamically for AC3, thus making lighter sounds play louder and compressing the overall range.
Also, Valex’s AC3filter will allow you to manually adjust the master gain, DRC and has a built-in equalizer as well. If your passing the sound over SPDIF then the majority of the filters settings will have no effect in order to pass the AC3 stream to your amp properly. Remember, the AC3 packets can not be altered or your amp would not detect the stream properly. Valex’s filter of course complies with this requirement.

12. Does the muxing software and settings affect AC3 playback?

If you plan to pass the sound through SPDIF and use AVI then the answer here is clearly yes. Thanks to the great AVI muxing software (AVImux GUI) from alexnoe, perfectly smooth playback with AVI/AC3 can be achieved easily. AVImux GUI is highly recommended to mux AC3 into the AVI container. Until now we were pretty much limited to using NanDub for AVI muxing but recently it was discovered that the muxing code in NanDub is serious flawed. Therefore, it’s no longer advised to use NanDub for any type of muxing. Now that we have AVImux GUI, in general we no longer have to worry about the interleave values and just use the default 500/75 provided. Please read the forum threads and the AVImux documentation for more details. You can find the latest builds on alexnoe’s home page here or grab it from Doom9’s download section. If you're using CD media to play back from and the CD drive is being accessed excessively causing playback problems then you might want to try an AVI file using AVImux. AVImux will cut down on the CD drive access considerably thanks to the nice “rec-list” feature implemented. VirtualDubMod is also an excellent choice for AC3 in AVI muxing.

Additionally, using a modern container format such as OGM or Matroska will cut down on muxing problems, these formats generally have no problems muxing AC3. For matroska muxing try MKVmerge or for OGM muxing (OggMux_Nic).

[Use always MKV]

13. What effects might I see if the AC3 files have been muxed improperly?

The most common problem seen if broken muxing software (NanDub) was used is to have jerky pan & scan scenes during playback or audio stutters. Another common problem might be excessive CD drive activity as mentioned above. Watch a movie closely when the camera pans from left to right or vice versa, it should be very smooth. An easy way to compare is to watch your ripped AVI without any audio muxed and then watch the same clip after you have muxed the AC3, you should not see any difference in regards to smooth playback (ie. Pan & Scan scenes). One last common symptom of a poorly muxed file is to hear the audio stuttering or dropping out during play, especially during high action scenes.

14. Why doesn't WinAmp playback AC3 files properly?

WinAmp doesn't support AC3 files natively. Valex has once again come to our rescue and provided an excellent plugin though. The AC3 plugin can be found here.

15. How can I playback a Dolby Digital wav (ddwav) on my PC?

VideoLan's VLC player version 0.7.1 and up supports playback of DDWavs and Dolby Digital-CDs.

Alternatively, you could first extract the 5.1AC3 stream from the DD-WAV using BeSplit and then play it in any player that is capable of playing ac3. An example command line is:

BeSplit -core( -input live.wav -output e:\track01.ac3 -type ddwav -fix )

16. What troubleshooting can I do to fix S/PDIF playback?

If you're still having problems troubleshooting, you can render the file in GraphEdit to get an idea on which filters might be needed. In GE, select the File Menu->render media file>load GraphEdit, add a File Source (Async) DirectShow filter, select your AVI file and to render it (right click on Output / Render Pin) to see what happens. That's how you'll know if DirectShow isn't able to properly render the file.
It’s been reported that Cineplayer's audio filter provides better SPDIF playback compared to say Intervideo. Cineplayer’s filter is also reported to work with SoundBlaster Live 5.1 cards with little effort. Valex’s AC3 filter is still preferable due to it’s ease of use and many features. Disable or uninstall the Morgan Stream Switcher filter. This filter can cause playback problems (e.g. stutter/skipping). The filter name is mmswitch.ax. Try the following command in the directory where the file is located- regsvr32 /u mmswitch.ax

If you're running Windows 2000, make sure you have at least Service Pack 2 and the latest sound card drivers are installed. SP2 solves some SPDIF out issues.

DirectX 8.1 and up is highly recommended. WinXP should have 8.1 already installed.

17. Any tips for Soundblaster sound cards?

The original Live Value(CT4780) cards requires a “stereo” 3.5mm mini plug and the newer models require a "mono” cable. If your sound card has an optical/Toslink or coax port, this will provide the most optimal quality, this compared to the multi channel analog ports. It’s possible that the coax can get interference known as EMI, although this is very rare.

On the SoundBlaster Live card software, make sure “Bypass SPDIF” option is _not_ checked.

Windows 9x/ME: In the SB mixer, check "digital out only". Also, uncheck "decoded AC3" and uncheck "bass redirection". Start your favorite media player and select SPDIF out in the audio configuration.

Windows 2000: At least install service pack 2 or later. Install SoundBlaster drivers v3512. In the SB mixer, check "digital out only". Also, uncheck "decode AC3" and uncheck "bass redirection” Start your favorite media player and select SPDIF out in the audio configuration.

Windows XP: This OS should not require any tweaking, driver shuffles or service packs for proper playback and SPDIF setup.

The Creative LiveWare software 5.1 sound test will not work for all 6 channels. It’s best to test your connection with a Dolby Digital 5.1 DVD if possible.

If you currently own a SoundBlaster 5.1 or a Live! model and are using SPDIF, your advised to buy the Hoontech adapter which allows correct voltage out for SPDIF. Link- (Important when using an IDE cable: The colored side of the cable should be on pin 1 on both Hoontech and
SoundBlaster card!)

Also, see the Creative FAQ for the 5.1 sound cards- http://www.soundblaster.com/support/...ily=1#faqstart

Creative’s Audigy can not decode DTS but it can pass it via SPDIF. The card can pass both DD and DTS as well as decode Dolby Digital also. It was pointed out by Goa333 that it might be a good idea to do a listening test between your AC3 decoding options such as WinDVD, PowerDVD, Sound Blaster AC3 hardware decoding and of course an external AC3 decoder. Your quality might be much better using one or the other. It has been reported that the Audigy card might suffer with AV sync problems using Wave Out as the audio device. Instead of setting this to “Wave Out” within the software player, it’s possible "DirectSound SBAudigy Wave Device" might solve sync problems if experienced.

18. What sound cards are most recommended for S/PDIF playback?

Audio Excel-Theater Excel-$19 USD
Hercules Fortissimo I & II-$45 USD
Hercules Game Theater XP- $95 USD
Turtle Beach Santa Cruz- $60 USD
Terratec DMX XFire 1024-$58 USD
Philips Acoustic Edge - $55 USD
M-Audio Audiophile 24/96 - $??(Highly recommended for Audiophiles)

Any C-Media 8738 based sound card has given great results whether it’s built-in to the motherboard or a PCI card. These cards are low budget but perfect for an inexpensive SPDIF setup. They also pass both DTS and DD without issue. Here are a few cards with the chipset-

AudioExcel MD-Mate - $??
Yamaha X-Wave 7000 Pro - $??
Zoltrix Nightingale Pro 6 - $35 USD

Philips Acoustic Edge is reported to work very well in VIA motherboards, perhaps more so than other cards. It’s advised the sound card be installed to the farthest PCI slot from the CPU.

As a last thought, the nForce and nForce2 motherboards seem to be perfect for Home Theater use

Note: All Creative cards are purposely not listed above due to the ongoing hardware/software issues on most of the models. Issues range from bad drivers, jitter, SNR to out of voltage specifications which could damage your external amp. The intent was to show cost effective sound cards which are easy to configure for multi-channel and SPDIF playback.

[Now all PC's have HDMI or SPDIF output without need specific sound card. That info is outdated]

19. Any AC3/spdif information for linux?

All the popular players for linux (xine, mplayer, vlc) include AC3 decoding support 'out of the box'. For s/pdif output, your player will need must also be ALSA (advanced linux sound architecture) compatible. Steps to using alsa are as follows:

1. Make sure your sound card is supported by checking alsa-project.org.

2. Make sure your kernel has alsa support compiled in. ALSA is the default sound architecture in the 2.6.x kernels, and many 2.4 kernels shipping with distributions are compiled with alsa built-in.

3. Install software & set up your soundcard: make sure you have the alsa-driver, alsa-lib and alsa-utils installed (may be called something else in different distributions) and run alsaconf. You will be prompted to select your soundcard. If you have no error messages, then run alsasmixer to set your default levels. Make sure to activate the S/PDIF output if you wish to use it. On my card, this is referred to as "IEC958 output".

4. Install a player with alsa support. The simplest way to see if mplayer has alsa support is to type 'mplayer -ao help' and see if alsa is one of the choices. If your mplayer does not contain alsa support, simply build mplayer from source, and if you have alsa-lib (+ alsa-lib-devel on some platforms) mplayer will detect and compile in alsa audio out support. Then add a '-ao alsa' to your playback command or your mplayer.conf file, and the alsa output will be used. To playback AC3 over spdif, you'll also need to add a '-ac=hwac3' switch to your commandline. Alternatively you can add 'ac=hwac3,' to your mplayer.conf file (note the comma after hwac3), which will cause mplayer to try ac3 pass through first on any file and then other codecs.

(http://www.linuxdvb.tv/documentation/ac3sound.html)
(http://www.au.linuxvideo.org/)
http://liba52.sourceforge.net/

20. Is it possible to record AC3 via SPDIF in?

Yes. For a guide and discussion, see this post and the comments that follow.

Last edited by tebasuna51; 27th September 2021 at 11:45. Reason: update
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DTS and DTS-HD FAQ

1. What is DTS?

DTS is a multi-channel home theater lossy encoded audio format developed by Digital Theater Systems. DTS as we know it on DVDs and LaserDiscs is not lossless, but is encoded at very high bitrates, typically 768-1536kbps.

[Like we can see in wikipedia there are new DTS formats for HD and 3D surround used in BluRay]

2. Is it possible to rip / play back DTS tracks from standard DVD discs and why would I want to do this?

[Now ffmpeg can decode DTS-HD and encode DTS standard, players like MPC-HC or VLC can decode or passtrough DTS-HD and can be muxed in MKV without problems.
Forget any other software mentioned here.
To rip DVD's or also BD's you can use MakeMKV, after you can use eac3to or ffmpeg to convert the DTS's like you want.]

Yes, a few ripping tools such as VOBrator, VobEdit, Smartripper, and DVD2AVI can demux DTS streams. DVDecrypter can be used to rip the VOB files with the DTS track from the DVD. Most people are using DVD2AVI to demux the DTS track. Note that it is not recommended to use vStrip to demux DTS due to the fact that vStrip often produces corrupt DTS streams by modifying the header.

For playback of DTS streams, the traditional method was to use the DTS audio decoder included with WinDVD. Gabest has written a filter that allows you to access the WinDVD decoder from many directshow players. You can download the DTS playback filters made available by Gabest (VobSub author) from Doom9’s download section.

However, thanks to the work of Gildas Bazin, there is now a GPL licensed DTS decoder library available - dtsdec / libdts. DTSDec has been implemented into a commandline decoder, and into Media Player Classic, foobar2000, and VideoLan's VLC. Although libdts is still in the early stages of development, either of these projects should be an excellent choice for free DTS playback.

Several tools can be used to mux DTS into popular containers. Later builds of AVImux GUI are capable of muxing DTS tracks into an AVI. DTS can also be muxed into Matroska files using MKVtoolnix.

There is (a guide available) on creating AVI or OGM with DTS inside courtesy of Wilbert.

Some users consider DTS audio to have a much fuller and richer sound compared to Dolby Digital. The only downside to using DTS files in our rips would be the file size. DTS files are typically around 768kpbs and up to 1536kbps, compare this to a Dolby Digital file which are commonly 384kbps or 448kbps. So as you can see by the bitrate difference, DTS files can be very large. However, a few very good uses come to mind for using the DTS tracks besides muxing them into a container in their raw format. In theory, transcoding a DTS file to an Ogg Vorbis or MP3 file could provide a better quality destination because of the source being so clean.


3. How can I transcode a DTS file?

Azidts can be used to transcode DTS with BeSweet (Note that you'll need WinDVD Platinum installed) or with one of the BeSweet GUIs (eg. OagMachine). The (Hypercube Transcoder) supports DTS transcoding as well. One could also use DTSdec (assuming a large amount of HD space) to decode to wav and then encode to any desired format.

Recent versions of foobar2000 support DTS playback and transcoding to any of fb2k's many output formats. Thanks to Kurtnoise there is an easy guide on transcoding DTS with fb2k to a variety of formats.

[eac3to or ffmpeg can do the job without problems also with DTS-HD]

4. Is it possible to encode DTS with BeSweet?

Yes, but you'll also need Surcode CD/DVD Pro, which encodes DTS. Please see (Q32 of the Besweet FAQ).

[ffmpeg can encode to dts standard for free.]

5. Is it possible to pass DTS CD Audio (music) over SPDIF?

The only player that currently supports this is VLC.

Another solution that has proven to work is to use the 2 pin digital output of your CD/DVD-ROM and connect it directly to the digital input of your receiver/amp. This method will bypass your sound card all together, sound cards might alter the stream and render the audio useless so the bypass is preferable. Keep in mind that your sound card should be capable of passing compliant Redbook audio. DTS 5.1 Audio CDs might not work using the SPDIF digital out port if your sound card up-samples the stream to 48kHz (SoundBlasters upsample). Unfortunately it seems most complaint WDM drivers are not capable of this for the moment but if your running Win9x then the VxD drivers should work. However, the M-Audio 24/96 or Delta Dio are reported to play the DTS audio over SPDIF perfectly.

6. I have encoded a 5.1 DTS WAV with SurCode DTS-CD. how do i playback it on my pc ?

VideoLan's VLC player supports playback of DTS-Wavs and DTS-CDs as of version 0.7.1.

If you have WinDVD platinum, install hypercube's DTSWAV filter, then you can use any directshow player (eg. windows media-player). Note that to use the DTSWAV filter, you must rename the files from *.wav to *.dtswav. If you have PowerDVD or any other DTS player, you first need to use BeSplit/BeSliced in order to extract the 5.1DTS stream out of the DTS-WAV.

[DTS-WAV files only make sense to burn a CD, is always better extract the DTS file with BeSplit with less size and better player support.]

7. What about playing DTS over spdif under linux?

Due to recent developments with libdts and alsa, this should now be fully working. First, follow the alsa installation steps described in Q19 of the AC3 FAQ, and make sure your spdif output is activated.

You'll also need a player with dts support. Both VLC and mplayer can be compiled for this if you have libdts installed. Then simply add a '-ac hwdts' to your mplayer commandline, or activate the spdif output option in the sound options of VLC.

8. What is the difference between DTS bitrates 1536, 1510 and 1509?

There are two kind of bitrates for DTS.
(quotations from TS 102 114 - V1.2.1 - DTS Coherent Acoustics; Core and Extensions)

First one is the nominal or "Targeted BitRate" and can be know with the field header "RATE":
"RATE specifies the targeted transmission data rate for the current frame of audio (see table 5.7)"
Valid values: 32..640,768,960..1472,1536,1920..3840

But, "Due to the limitations of the transmission medium the actual bit rate may be slightly different from the targeted bit rate, as listed in table 5.8"
Then there are a second one, the real or "Actual BitRate" that can be less (1510, 1509, ...) than the nominal (1536).

When you encode a .dts file the frames are padded with 0's from the "Actual BitRate" to the "Targeted BitRate".
If you encode to a .cpt (compact) file the frames aren't padded and the file size is less than the .dts (the audio stream is the same).
BTW, you can rename safely the .cpt to .dts without problems.

To know the "Actual BitRate" we need know other field header values:
"NBLKS Number of PCM Sample Blocks", typically 512.
"SFREQ Core Audio Sampling Frequency", typically 48000
"FSIZE Primary Frame Byte Size", not always the same (encoder choice), I see values like 2013 and 2012, then:

Actual Bitrate = 2013 x 8 x 48000 / 512 = 1509.75 Kb/s (rounded to 1510)
Actual Bitrate = 2012 x 8 x 48000 / 512 = 1509 Kb/s

A DTS with "Targeted BitRate" 1536 can have an "Actual BitRate" of 1510 or 1509.
Same differences can be aply to "Targeted BitRate" 768.

9. For what eac3to, when extract a DTS, show: "Patching bitdepth to 24 bits..."?

The DTS stream is not changed at all, only a metadata in the header is changed because some decoders can use that data (wrongly) to decode the stream to 16 bits instead 24 or better.

The problem is a wrong understanding of the metadata, it is not the bitdepth of the DTS (the lossy encodes don't have bitdepth, only lossless like DTS-MA), it is the "Source PCM Resolution" used to encode that DTS, but no matter the source resolution was 16 or any we never can recover the same precission with a lossy encoder.

The lossy decoders always work internally with float samples and downsample to 16 bits int it is not recommended at all, if we decode a lossy format is to recode to other format (for compatibility) or to edit, and then must preserve the better bitdepth than support the next encoder or editor.

eac3to by default always decode lossy formats to 24 bit int but using the undocumented parameter -full we can obtain until 64 bits float (with -libav decoder, not libdcadec).

Last edited by tebasuna51; 30th September 2021 at 13:01. Reason: update
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AAC FAQ

1. What is AAC?

AAC stands for Advanced Audio Coding which is a general purpose lossy audio compression format defined by the MPEG-4 standard and developed by Fraunhofer, Dolby, Sony and AT&T as major contributors. It is the natural successor of MPEG Layer III / MP3 in the new multimedia standard MPEG-4 that uses MP4 as the container format for all kinds of content. AAC is able to include 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 15 low frequency enhancement (LFE, limited to 120 Hz) channels, up to 15 data streams and much more

[At this moment is the more compatible audio format for stereo audio.
For multichannel you need conexion by HDMI to your AVR.
Only Opus have better ratio quality/size but is still less compatible.
That make to AAC the recommended lossy format to recode when we need save space or after a needed edition.]


2. What are HE-AAC and LC-AAC?

Both are in the MPEG4 AAC standard. LC means "low complexity" and HE means "high efficiency". HE-AAC is also called AAC SBR/AAC+/aacplus, etc. Note that HE-AAC is meant to be used for low bitrate encoding and it is great for multichannel files (smaller filesize). For HE-AAC files, a HE capable decoder is required as well, see Q6. If you want to see comparison with other popular audio codecs at 64kbps, look here.

More technically, HE-AAC is the combination of AAC LC and SBR. SBR stands for Spectral Band Replication. The point of SBR is to encode at low bitrate the entire frequency range without introducing additional artifacts. Traditionally, at low bitrates, audio codecs had the choice of reducing the frequency range and downsampling (see the MP3 FAQ #7) or introducing unpleasant audio artifacts. SBR fixes this problem by leaving the underlying codec to code the low range of frequencies, and the SBR decoder generates the high range frequencies through analysis of the low frequencies, along with guidance information embedded in the bitstream at a very low bitrate (~2 kbps). Therefore this is why using a non-SBR decoder is a very bad idea, as your bandwidth (frequency response) will be severly limited. It can also be seen why this is called Spectral Band Replication, as the frequencies are replicated, and not reconstructed.

Like with MPEG-4 Video, AAC comes in different Profiles, from which the Low Complexity (LC AAC) Profile (aka MAIN @ Level 2) is the one most widely used in the consumer market (for example in Apple's very popular iTunes music store)
other profiles are for example the Long Term Prediction Profile (LTP), Scalable Sampling Rate (SSR) or Low Delay (LD)

[Use HE only when you need a really low bitrate, is not recommended at all to preserve the audio quality.
You can see after the bitrate range to use it, also for HE v2 (LC+SBR+PS) adding Parametric Stereo]

3. How can I encode AAC?

[Read the Wiki with Actual encoders features

- To obtain the Apple AAC encoder:

1) Download the last qaac CLI interface, now v2.72
2) Download iTunes64Setup.exe maybe from here
3) If you do not want install iTunes you can extract only the needed files executing at same folder (you need have 7z installed) makeportable2.cmd
4) Put the folder QTfiles64 with qaac64 (or QTfiles with qaac.exe if your OS is only 32 bits)
5) To use less size you can replace icudt62.dll with the dummy one.
Like you can see here the encoder CoreAudioToolbox.dll is the same (v 7.10.9.0) from 2017-03-16.

- FhG AAC encoder, (Winamp 5.66)

- Fraunhofer FDK AAC encoder you can compile new commits or ask to other user.

- Nero AAC encoder

- FFmpeg 3.0+ AAC encoder already know.

All support STDIN input and for that can be used in 'piped' CLI and GUI's like BeHappy or UsEac3to

Only a bitrate comparative between VBR mode in all them.
Over a song: Here Comes My Baby (Cat Stevens) 2021-09-28
Code:
                  Fraunhofer   Fraunhofer
  iTunes Qaac       WinAmp      Android      NeroAacEnc        ffmpeg
 * V  Avg/Max    FHG Avg/Max  FDK Avg/Max  qualit Avg/Max    q   Avg/Max
---------------  -----------  -----------  --------------   ------------
104-96  232/362                5  230/451  .65  234/329     1.2  230/601
 95-87  198/327                            .56  194/271     1.1  208/559
 86-78  161/261  5 177/287                 .47  157/229     0.9  161/556
 77-69  146/243                4  134/287  .44  142/217     0.8  145/559
 68-59  128/222  4 122/207                 .41  128/198     0.7  132/562
 58-50  114/207                3  109/254  .38  115/219     0,6  115/505
 49-41   99/191  3  96/175     2   93/224  .35  101/177     0.5  103/445
 40-32   82/174                1   86/219  .32  90/163      0.45  94/412
 31-23   77/150                            .30  84/125 HE   0.4   83/321
 22-14   68/149  2 66/114 HE               .26  70/107 HE   0.35  71/296
 13-5    59/134                            .23  60/93  HE   0.3   55/191
                 1 36/75 HEv2              .15  33/48 HEv2

* The V parameter have a range value, is the same put 50 than 58, and the encoder can show TVBR 54
At this moment seems the ffmpeg quality mode still is under changes (I see variations with previous versions), the Max. bitrates seems too big compared with the others.

Of course the recommended now is the Apple iTunes Qaac using CoreAudioToolbox v7.10.9.0.

The next links are deprecated.]

3b. What about encoding AAC under other platforms (Linux, MAC OS 10, etc.)?

FAAC and FAAD, being GPL software, can be built under OS X, Linux, and most other unixes. The FAAC codec is also implemented in avidemux, mencoder,gstreamer, and other encoding tools. Most major multimedia-capable distributions include these packages.

General FAAC compiling instructions are as follows: (taken from the FAAC documentation)

1. Make sure you have autoconf, automake and libtool installed. For MP4 support, you must have libmp4v2 (included in older source of faad2, or recent cvs source of FAAC) installed.
2. Unpack the source archive: tar xvzf faac-x.xx.tar.gz
2. cd to FAAC source dir
3. Run: ./bootstrap
./configure
make
make install

Binary packages are also availble for several platforms:
Debian
RPMs

4. Is it possible to convert AC3/DTS to AAC and preserve 5.1?

[Also TrueHD/DTS-HD with the best encoder in ratio quality/size, but remember than your player must be conected by HDMI to your AVR.
Use CLI or GUI's like BeHappy or UsEac3to]

(Yes, it is possible with Oagmachine/BeSweetGuI. Doom9 has a guide available. However, keep in mind that AC3 and DTS are lossy formats, and any transcoding of these sources will result in a significant reduction of quality. HE-AAC is recommended for 5.1 encodes when a great reduction of size is desired. 5.1 transcoding is also possible using the 3ivx filter suite along with graphedit (AC3->AC3 parser->AC3 filter->3ivx audio encoder->3ivx media muxer->MP4) or foobar2000. For more information on AC3 and DTS transcoding see the respective FAQs.)

5. Is there an opensource AAC encoder/decoder available?

[ffmpeg]

FAAC and FAAD2 are an open source AAC encoder (LC only) and decoder respectively. The CoreAAC Directshow filter, winamp plugin, and many other AAC decoders are derived from FAAD2 which has the abillity to decode HE-AAC as well as LC. The official website of FAAC/FAAD2 is (www.audiocoding.com.)

6. How can I mux AAC into AVI/OGM/MKV/MP4?

[Forget AVI and OGM, use only MKV or MP4]

AVI: AVI MUX GUI is the only application that currently allows AAC muxing in avi. Note that AVI is not recommended for AAC as modern containers such as MP4 (which is AAC's native container).

OGM: Ogg Mux Nic can be used for direct muxing of LC AAC into OGM. For HE-AAC, use the 3ivx filter suite and Graphedit.

MKV: MKVtoolnix and AVI MUX GUI can both be used to mux AAC in Matroska.

(MP4: 3ivx filter suite and Graphedit.
Nero Recode2 automates direct transcoding of DvD to NeroDigital and HE AAC in the MP4 container. Other tools that support MP4 muxing include Ivan&Menno and mp4UI.) For more information on the MP4 (MPEG-4) container format see the MP4 FAQ.

[To mux AAC in MP4 you can use ffmpeg or Yamb]

Note that when muxing HE-AAC from raw .aac streams, you have to indicate to the muxer that "AAC is SBR" in MKVtoolnix and AVImux GUI. When muxing .mp4, SBR is automatically detected.

7. What do I need for AAC playback?

[A player like Foobar2000, MPC-HC or VLC, nothing more. If you need DirectShow filters use LAV Filters]

Download and install the CoreAAC DirectShow filter or the 3ivx audio decoder (bundled with 3ivx codec), both are excellent for AAC (Both LC and HE) playback in any directshow compatible media player. For the latest CoreAAC filter and winamp/foobar plugins, refer to john33's thread at HydrogenAudio.org.

Note that for MP4 streams, 3ivx or another MPEG-4 splitter is required for playback in DirectShow compatible media players. For raw AAC streams, the AACparser filter from rarewares is required. When muxing in OGM or Matroska, an appropriate splitter is needed as well. See the FAQs in the New Container Formats forum for more information.

7B. What about playback of AAC under linux / *nix?

Any relatively recent build of mplayer includes both MP4 file format support and AAC decoding via FAAD. FAAD is included in the mplayer source code, and should require no additional libraries or compilation.

VLC supports AAC and MP4 as well, but if you build VLC from source you'll need to have FAAD installed for AAC support. As always you should check through the VLC required libraries list before building VLC from source.

8. What's the difference between MPEG-4 and MPEG-2 AAC?

There is practically no difference between MPEG-2 and MPEG-4 AAC streams except in the header. But there are some applications like QuickTime which don't accept MPEG-2 AAC. Use (Ivan & Menno) to change the header. Other programs capable of the header fix are Nic's (AAC Patch) or foobar2000.

9. What is the difference between *.AAC and *.MP4 and *.M4A?

AAC files usually contain AAC with ADTS headers or raw AAC data streams. Raw AAC cannot be processed without decoding some of the stream. MP4, on the other hand, is a container format of the MPEG4 standard which can contain AAC streams and many other things. The data format of the two is very different and hence you cannot rename them to one another, they must be muxed (put into the container) or demuxed. Muxing of AAC into MP4 and demuxing can be done with Ivan & Menno. M4A is normally audio in an MP4 container that has been renamed to clarify that it is audio only. Itunes, Winamp5, and Realplayer gold encode to M4A now by default. Note that M4A and MP4 files can be switched to each other by simply renaming, since they are both extensions for the same container format.

Note that the "Export ISO 13818-7 Stream" option in the Nero encoder produces *.AAC files with ADTS headers. Again this is not a raw aac stream, (Ivan & Menno) is required to convert to raw aac.

More information on the MP4 container format can be found in the appropriate FAQ.

Last edited by tebasuna51; 1st October 2021 at 10:15. Reason: Update
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10. How can I split and join AAC files?

AAC in MP4:
MP4Box can be used for splitting and joining all MP4 files including those with AAC audio. See this thread for full information.

MP4 with AAC inside can also be edited by QuickTime Pro.

Raw AAC streams:
Raw aac streams (*.aac) can be split and joined by using BeSplit.

Splitting:
Code:
besplit -core( -input in.aac -prefix out -type aac -ending ) -split( 0.0 x.x )
This command-line would cut in.aac into two parts: 0.0 s up to x.x s and x.x s up to the end of the file (-ending).

Joining:
First create a list file, which is a text file listing each file to be joined on a separate line. Then save as join.lst and use BeSplit:
Code:
besplit -core( -input join.lst -prefix out -type aac -join )
AAC streams (*.aac) can also be joined via command prompt:
Code:
copy /b first.aac + second.aac joined.aac
11. What are the pros and cons of the AAC audio format?

Pros:
  • Low bitrate (with comparable quality to other codecs) and hence low filesize, with SBR in specifications.
  • Multichannel (up to 48 full bandwidth channels can be implemented)
  • AAC is the official audio format of the MPEG-4 standard, and future hardware support is very likely.
Cons:
  • (No standalones that playback AAC are available at this time.) [All devices play fine AAC now, but must have HDMI conexion to AVR's for multichannel]
  • AAC is heavily patented technology.

12. Is it possible to pass AAC over S/PDIF?

[Still not, but use HDMI to pass PCM multichannel]

It is possible, although it has not yet been fully implemented. Only a few hardware devices currently support AAC decoding, and no DirectShow filter has yet been implemented with AAC pass-through ability. However, this is definitely a possibility for the future. Read more about S/PDIF and AAC hardware support here and here.

However, until AAC over SPDIF becomes more viable, there are several DirectShow filters that can be used to encode six channel AAC to AC3 over S/PDIF on the fly. Dolby Out DS filter by Haaan , as well as 1.x versions of AC3Filter can be used for this purpose. Although this reencoding on the fly process will theoretically result in a quality loss, using the full 640 kb/s bitrate allowed will result in an excellent listening experience and minimal quality loss.

13. How are bitrates divided among channels in 5.1 LC and 5.1 HE AAC? What bitrate should be used for 5.1 AAC to be equvalent in bits/channel to 128kbps?

For the bits/channel to be the same, one must add 2x(bitrate per stereo)(FL,FR,RL & RR)+ 1x(mono bitrate(Centre))+ a small additional bitrate for lfe. Therefore the total bitrate should be about 2.5-3x as much as stereo to achieve the same bits/channel. So to achieve the datarate equivalent to 128kbps stereo: 128kbps*2.5=320 kbps. 320kbps is a good quality bitrate for LC AAC. However due to the efficiency and channel coupling of AAC, one can encode HE-AAC at 128kbps and still have decent quality and 5.1 channels.

14. At what bitrates is the HE-AAC profile used in Nero?

[NeroAacEnc output AAC-HE at -q 0.31
The rest of FAQ's are usseless now.]

For CBR settings, 96kbps and below use HE-AAC. For VBR settings, streaming profile and below can produce HE-AAC. If your input source is 5.1 aiff , then HE AAC can be produced at 128kbps using Nero only.

15. What gain setting should I use to encode AAC with BSN/Oagmachine?

Eventhough BeSweet/Bsn has implemented Hybrid gain, there are no Directshow filters capable of using the hybrid gain information at this time. So it is not recommended at the moment. Instead use normalization or pregain.

16. Is there any AAC ACM codec that can be used with VirtualDub?

There is currently no ACM AAC codec due to AAC and AVI not being completely compatible. In the FAAC project this was once attempted and never finished.

17. Encoding doesn't start with Besweet/Nero, i have installed all the BeSweet and Bsn files. Please help?

For BeSweet/BSN/Oagmachine/BeSweet GUI to encode Nero AAC, three dlls from a valid nero installation are needed: 'aac.dll', 'aacenc32.dll' and 'NeroIPP.dll'. These dlls need to be put in the BeSweet folder. From version 60023 onwards aac.dll and aacenc32.dll check for the NeroIPP dll in the registry, therefore a valid Nero Ultra Edition or NeroMix installation is needed.

18. I cannot see the Nero encoding GUI when encoding with OagMachine/BeSweet, how is it activated?

In Oagmachine, under the AAC encoding option, check "show configuration dialogue". If you are using command line, then add a '-config' to the '-bsn' section.

19. Why is my ac3/dts encoded at 48Khz in Oagmachine/BeSweet GUI slow when played back?

That's because you're probably using an old build of BeSweet, since the bug had been fixed in bsn.dll v0.21 (bundled with BeSweet). Check your BeSweet logfiles to make sure you have bsn.dll v0.21.
More info on the old bug :
The Nero encoder expects specific sample-rates depending on the bitrate, profile, and number of channels. When encoding using BSN earlier than v0.21, the internal nero resampler is not used and you must feed the Nero encoder with the correct sample rate. See this thread to determine your correct sampling rate and a more thorough explanation by bond.

20. How can I ensure my encoded AAC is HE and not LC?

Use the CoreAAC DirectShow filter properties to check "AAC+SBR". Foobar2000 also shows the same.
In the 3ivx audio decoder config it is shown as HE-AAC.

21. What switch for multichannel should be used in Oagmachine/BeSweetGUI when using the 600X nero plugin?

Use -6chold for nero plugins earlier than or equal to v6.0011(old). Above 6.0011 use -6chnew.

22. My Nero encoded HE-AAC contains popping noises, what did I do wrong?

This problem occurs when you enable PNS in the Nero encoding GUI. Uncheck the PNS option to get rid of the problem.

23. Are there any recommendations for encoding with good quality with FAAC?

Yes, see the FAAC wiki and also this discussion on FAAC 5.1 encoding.

24. How can I convert AAC to WAV or AC3?

First you should read both this document and the AC3 FAQ to learn about the concepts and tools available. However if you still need a walkthrough, see this guide or this graphedit guide by magicclue.

25. Where can I find more information about AAC?

Audiocoding Wiki
Coding Technologies
Dolby

Last edited by tebasuna51; 27th September 2021 at 03:00. Reason: Update
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Credits:

A big thanks goes to all the people mentioned below, their contributions to the many parts of the FAQ and their general forum help is invaluable.

Doom9 for running a great site and forum

DSPguru
pacohaas
MaTTeR
bond
Tuning
tiki4
rjamorim
tangent
Wilbert
akan314
hans-jürgen

All for their excellent contributions.


Reference websites:

http://www.audiocoding.com
http://www.hydrogenaudio.org
http://www.dolby.com
http://www.mp3dev.org

Changelog:
1.7.04: Initial reorganization and posting
1.8.04: Updated DTS Q2 regarding vStrip. Linked to Wilbert's guide. Clarified Q4 regarding Surcode Pro & Q6 regarding DTSWAV filter.
1.8.04: Reorganized AAC Q3 for clarity.
1.8.04: Updated General Q16 winamp plugins.
1.9.04: Updated AAC Q10 with command prompt joining method.
2.9.04: Updated DTS Q2 for misleading quality information.
2.23.04: Updated Vorbis Q2 and AAC Q12 with Dolby Out DS information.
3.2.04: Updated AC3 Q15 and DTS Q5 & Q6 with information about using VLC for DD/DTS-Wav/CD playback.
3.11.04: Updated rarewares links throughout FAQ due to domain change.
3.11.04: Updated AAC Q3 regarding Real AAC encoding ability.
4.5.04: Updated AAC Q3 links, updated AAC Q9 formats info, merged AAC Q23 to AAC Q9 (M4A)
4.5.04: DTS Q4 clarified regarding Surcode Pro.
4.5.04: AC3 Q2 updated re ac3enc.
6.27.04: DTS FAQ updated re DTSDec/libdts.
6.27.04: Vorbis Q2 updated with vorbis encoders.
6.27.04: AC3Filter 1.0x information updated in several questions.
9.4.04: AAC FAQ updated re: FAAC: new url, new plugins, *nix build information added.
9.6.04: Added linux questions and information across the board: MP3 Q11, Vorbis Q8, AC3 Q2B & Q19, DTS Q7, and AAC 7B.
9.7.04: Added winLame & HeadAC3he (FAAC) information to AAC Q2.
6.19.05: AAC FAQ additions: Added bond's AAC sticky info, added VBR QT info, added magicclue's guide links.
6.22.05: Added MP4Box splitting to AAC Q10.
10.12.05: Updated DVD-audio information in general, bumped recommended lame version in MP3 section, updated vorbis versions / download links.
2010.10.13 Added FAQ #8 DTS bitrates (tebasuna51)
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KpeX
Audio FAQs: General | BeSweet | SVCD/MP2 | MP3 | Vorbis | AC3 | DTS | AAC
Linux Audio/Video FAQ

Last edited by tebasuna51; 13th October 2010 at 13:21.
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