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17th May 2007, 12:08 | #581 | Link | |
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Quote:
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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22nd May 2007, 17:55 | #582 | Link |
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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% |
22nd May 2007, 19:30 | #583 | Link |
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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. |
22nd May 2007, 19:52 | #585 | Link | |
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Quote:
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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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% |
22nd May 2007, 21:55 | #587 | Link |
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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 |
27th May 2007, 12:34 | #588 | Link | |||
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Quote:
Quote:
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I used word "Approximate Bitrate" , not "Bitrate". We are using "Q mode" |
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27th May 2007, 13:24 | #589 | Link | |
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Quote:
"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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16th June 2007, 21:43 | #591 | Link | |
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Quote:
Code:
Channels=6, BitsPerSample=16 int, SampleRate=48000Hz oggenc2.exe -Q --quality 3 -o "G:\Test.ogg" - |
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21st June 2007, 18:08 | #592 | Link | |
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Quote:
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. |
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22nd June 2007, 00:56 | #593 | Link |
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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 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. ... |
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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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! |
3rd July 2007, 18:40 | #598 | Link | |
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3rd July 2007, 19:00 | #599 | Link | |
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Quote:
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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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:
Last edited by jordisound; 3rd July 2007 at 23:46. |
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