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Old 17th May 2007, 12:08   #581  |  Link
tebasuna51
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Quote:
Originally Posted by thuongshoo View Post
@Tebasuna51: I tested 2 file and feel Ok. Everyone please re-test
I say the approximate bitrate is only for stereo audio:

StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.
StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s

I propose only a literal change:
Variable Bitrate Q=3 (approx. 112 kb/s for stereo)
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Old 22nd May 2007, 17:55   #582  |  Link
MuLTiTaSK
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.mpa -> .ac3 conversion

this tool is amazing i mostly use it to convert .ac3 files i extract from my dvd's to .mp3 format and it works great

but now i want to use it to convert a 224Kbps CBR 2CH Stereo 48KHz .mpa i extracted with DGIndex from one of my tuner caps to .ac3 format so i can import it into a dvd authoring program

i never did this before so i'am not sure what encoder to use and what settings to set in BeHappy i tried with ffmpeg but not sure if i did it right

i'am using lastest Shon3i BeHappy package with 2007-04-01 BeHappy mod http://forum.doom9.org/showthread.ph...715#post998715

below is the log i got after the encode was done
hope someone can help me do this conversion right thanks in advance

settings i used in BeHappy



Code:
Starting job test T01 DELAY 0ms.mpa->test T01 DELAY 0ms.ac3
Found Audio Stream
Channels=2, BitsPerSample=32 int, SampleRate=48000Hz
ffmpeg.exe -i - -y -acodec ac3 -ab 192 "H:\Caps\test T01 DELAY 0ms.ac3"
Writing RIFF header to encoder's StdIn
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete
#### Encoder StdErr ####
FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --enable-mingw32 --enable-memalign-hack --enable-gpl --enable-a52 --enable-dts --enable-mp3lame --enable-faac --enable-amr_nb --enable-faad --enable-amr_wb --enable-pp --enable-x264 --enable-xvid --enable-theora --enable-libogg --enable-vorbis 
  libavutil version: 49.0.0
  libavcodec version: 51.9.0
  libavformat version: 50.4.0
  built on May 13 2006 18:31:30, gcc: 4.1.0 [Sherpya]
Input #0, wav, from 'pipe:':
  Duration: N/A, bitrate: 3072 kb/s
  Stream #0.0: Audio: pcm_s32le, 48000 Hz, stereo, 3072 kb/s
Output #0, ac3, to 'H:\Caps\test T01 DELAY 0ms.ac3':
  Stream #0.0: Audio: ac3, 48000 Hz, stereo, 192 kb/s
Stream mapping:
  Stream #0.0 -> #0.0

video:0kB audio:145896kB global headers:0kB muxing overhead 0.000000%
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Old 22nd May 2007, 19:30   #583  |  Link
tebasuna51
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@MuLTiTaSK
What is the problem?
Seems the conversion is ok. You have a 'H:\Caps\test T01 DELAY 0ms.ac3' about 145896 kB.
Don't worry about info from ffmpeg, all is ok.

Maybe you can use the last Aften like encoder with more options than ffmpeg.

Last edited by tebasuna51; 22nd May 2007 at 19:33.
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Old 22nd May 2007, 19:38   #584  |  Link
shon3i
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yep, everythig is normal like log says, if file is playable, there is nothing to do.

btw new package will be redy, at the end of this week
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Old 22nd May 2007, 19:52   #585  |  Link
MuLTiTaSK
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Quote:
Originally Posted by tebasuna51 View Post
@MuLTiTaSK
What is the problem?
Seems the conversion is ok. You have a 'H:\Caps\test T01 DELAY 0ms.ac3' about 145896 kB.
Don't worry about info from ffmpeg, all is ok.

Maybe you can use the last Aften like encoder with more options than ffmpeg.
i wasnt sure if BeHappy was reporting a error by looking at the log i was a little confused thank you guys for clearing that up for me

would i get better quality using aften to encode the .mpa to .ac3?

i notice it had more settings but i did'nt know which ones to set so i used ffmpeg ;/

would these settings be correct for my conversion thanks again
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Old 22nd May 2007, 20:05   #586  |  Link
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These default settings are ok, equivalents to ffmpeg.

Aften is based in same code than ffmpeg but have some improvements, maybe not noticeable but there are.

If you want ensure the max volume without clipping you can use the DSP function Normalize 100%
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Old 22nd May 2007, 21:55   #587  |  Link
MuLTiTaSK
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worked like a charm thanks alot tebasuna51 & everyone else keeping this great program alive

shon3i your packages help a great a deal thank you very much for building them saves new users to the program alot of grief

i'am still surprised why more people dont use this gem

is there a reason BeHappy is not listed on sites like?

http://www.digital-digest.com/softwa...tegory-23.html
http://www.videohelp.com/tools/sections/audio-encoders
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Old 27th May 2007, 12:34   #588  |  Link
thuongshoo
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Quote:
StereoTrack -> Ogg Q=3 approx. 112 kbs -> Real 113 Kb/s OK.
Quote:
StereoTrack -> Downmix to mono -> Ogg Q=3 approx. 112 kbs -> Real 62 Kb/s
62 = 112/2
Quote:
StereoTrack -> Upmix to 5.1 -> Ogg Q=3 approx. 112 kbs -> Real 441 Kb/s
I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"
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Old 27th May 2007, 13:24   #589  |  Link
tebasuna51
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Quote:
Originally Posted by thuongshoo View Post
I don't know how 5.1 file is encoded.
I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode"
5.1 (441 Kb/s) is encoded with Q mode and literal:

"Variable Bitrate Q = 3 approximate bitrate 112 kbs"

And I propose a literal like:

"Variable Bitrate Q = 3 (approx. 112 Kb/s for stereo)"
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Old 16th June 2007, 17:05   #590  |  Link
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are you sure? teba? that the 5.1 is encoded witht the Q mode?
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Old 16th June 2007, 21:43   #591  |  Link
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Quote:
Originally Posted by paranoid87 View Post
are you sure? teba? that the 5.1 is encoded witht the Q mode?
I'm sure about the commandline:

Code:
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz
oggenc2.exe -Q --quality 3 -o "G:\Test.ogg" -
and the result is a ogg 5.1 with bitrate 441 (Foobar Properties)
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Old 21st June 2007, 18:08   #592  |  Link
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Quote:
Originally Posted by tebasuna51 View Post
Last links resumed:
2007-03-24 Last Shon3i BeHappy package
I'm using this version, but encoding to flac doesn't manage:
Code:
Starting job audio.ac3->audio.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "F:\audio.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7172361216 -
Writing PCM data to encoder's StdIn
Error: System.IO.IOException: Using the pipe has finished.
source: ac3 , with NicAC3Source , filters: none, flac: best
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Old 22nd June 2007, 00:56   #593  |  Link
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Quote:
Originally Posted by chros View Post
I'm using this version, but encoding to flac doesn't manage...
Works for me with small samples (--input-size=183057408):
Code:
Starting job Test.ac3->Test.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "D:\Test.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=183057408 -
Writing PCM data to encoder's StdIn
Finalizing encoder
Complete
But also crash with big sources (--input-size=7068791808):
Code:
Starting job 136_min.ac3->136_min.flac
Found Audio Stream
Channels=6, BitsPerSample=24 int, SampleRate=48000Hz
flac.exe --best --force -o "G:\136_min.flac" --silent --force-raw-format --endian=little --channels=6 --bps=24 --sample-rate=48000 --sign=signed --input-size=7068791808 -
Error: System.ApplicationException: Can't start encoder: Cannot process request because the process (1096) has exited. ---> System.InvalidOperationException: Cannot process request because the process (1096) has exited.
...
The same source can be transcoded to wav or aac without problems, then seems a flac encoder problem.
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Old 22nd June 2007, 07:10   #594  |  Link
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i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input

BHH
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Old 22nd June 2007, 10:38   #595  |  Link
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Quote:
Originally Posted by buzzqw View Post
i can confirm the bug using ffmpeg for piping to faac.
sothing is wrong in faac pipe input
Are you talking about faac or flac?
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Old 24th June 2007, 18:12   #596  |  Link
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ops.. misread... i mean faac not flac

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Old 3rd July 2007, 15:07   #597  |  Link
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I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?
Thanxs!
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Old 3rd July 2007, 18:40   #598  |  Link
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Quote:
Originally Posted by jordisound View Post
I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?
Thanxs!
You have to feed your source using an AVS script and add the GetChannel command to grab each channel separately. Add each script to the queue and then run it to produce each wav file.
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Old 3rd July 2007, 19:00   #599  |  Link
tebasuna51
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Quote:
Originally Posted by jordisound View Post
I've tried to convert an ac3 4 channels from an old DVD to 4 wavmono using behappy (installer 20070315). After process only appears FL, FR.
Using besweet i've got the 4 channels FL, FR, C, S (SL, SR are the same, and LFE is empty.)
So, behappy is not able to convert 4.0 to 4 wavmono? or have I to change the configuration?
The problem is in the ac3 decoder (NicAudio.dll) and is detected in this thread

When the ac3 is 3/1 (FL, C, FR, S) NicAudio make an automatic downmix to:
3/1.0 -> stereo downmix
lt = 0,4xFL + 0,3xC + 0,3xS
rt = 0,4xFR + 0,3xC + 0,3xS

If the ac3 is 2/2 (FL, FR, SL, SR) NicAudio works ok.

- To decode properly an ac3 3/1.0, with free tools, you need use Azid.exe directly with:

azid.exe -d3/1 -ol,r,c,sl input.ac3 output.wav

The output.wav is a 4 channel (FL, FR, C, S) wav and you can split in mono wav's with WaveWizard or Wav2mono (1)

- Also you can use Foobar2000 to Convert to a WAVE_FORMAT_EXTENSIBLE wav or, if you use Wav2mono with Foobar, directly to 4 mono wav's (1)

(1) The wav's are named FL, FR, BL, BR and must be FL, FR, C, S
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Old 3rd July 2007, 22:17   #600  |  Link
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ok. i'll try and tell you if it works. thank you tebasuna.

edit:
it works.
Quote:
The wav's are named FL, FR, BL, BR and must be FL, FR, C, S
exact! this is very important!

Last edited by jordisound; 3rd July 2007 at 23:46.
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