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Old 27th December 2005, 13:45   #1  |  Link
dimzon
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BeHappy - AviSynth based audio transcoding tool (UPD 19-07-2006)



Use-Case
BeHappy is designed mostly for movie audiotrack transcoding. It's powerfull BeLight/BeSweet replacement. So do not ask me about:
  • Batch trancoding support
  • Tags support
If You need to transcode multiple files at "one click" and/or tagging support - use foobar2000 converter

Screenshots






















System requirements
Installation
  • Download and extract latest BeHappy from BeHappy Workspace
  • Download and extract latest NicAudio.dll from BeHappy Workspace, place NicAudio.dll to AviSynth plugins folder
  • If You plan to use Stereo To 5.1 UpMix functionality download and extract latest SoxFilter from here, place it to AviSynth plugins folder
  • If You plan to use AudX MP3 5.1 surround encoding download and extract latest enc_AudX_CLI from BeHappy Workspace, place enc_AudX_CLI.exe and audxlib.dll to BeHappy folder
  • If You plan to encode to MP3 download and extract Lame encoder from rarewares (recomended version is 3.97b2). Place lame.exe into BeHappy folder
  • If You plan to encode to OggVorbis download and extract fresh OggEnc2 from rarewares, place oggenc2.exe to BeHappy folder
  • If You plan to encode to FLAC download and extract fresh FLAC coder, place flac.exe into BeHappy folder
  • If You plan to encode to WavPack download and extract fresh WavPack encoder, place wavpack.exe into BeHappy folder
  • If You plan to use Coding Technologies AAC encoder
    • Download and extract latest enc_aacPlus from BeHappy Workspace, place enc_aacPlus.exe to BeHappy folder
    • Download and extract latest GPAC's MP4Box, place mp4box.exe to BeHappy folder
    • Download and install fresh WinAmp player (Full version is enought)
    • Copy enc_aacplus.dll and nscrt.dll from Winamp\Plugins to BeHappy folder
    • If You doesn't want to use WinAmp You can uninstall it now
  • If You plan to encode to Nero AAC download and extract fresh FREE NeroDigital AAC, place neroAacEnc.exe and neroAacEnc_SSE2.exe into BeHappy folder



Original idea


Workspace is hosted @

Note
MeGUI since v 0.2.3.2038 23 Jan 2006 share same code with BeHappy. So you can use MeGUI for audio encoding via AviSynth too. MeGUI audio part has less functionality/flexibility but is more stable

Mini FAQ:
TODO

Last edited by dimzon; 19th July 2006 at 01:07.
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Old 27th December 2005, 20:48   #2  |  Link
tebasuna51
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*** READ THIS: actualized 2008-02-05 ***

- BeHappy Workspace link don't work because GotDotNet Workspaces have been phased out.

- New BeHappy home: CodePlex BeHappy project with last mod's included.


------------------------------------------
*** Now the original post ***
------------------------------------------


Good job, Dimzon!

For me work the 'Delay', 'Split', 'Downmix', ...

Only with the 'Upmix' I get always:
Error: Can't find audio stream!

Please be patient with me, because I have a lot of comments:

Questions
In Avisynth internal plugin info say SuperEq(string filename), then wait for a parameter like this:
SuperEq("C:\Equalizer Presets\Loudness.feq")
Can work SuperEq(clip, values="0 0...96...0")?

In Upmix using Reverberation you put the plugin reverb(). Is a external plugin and need be downloaded?

EnsureVBRMP3Sync(). We can work with mp3 audio?. Is not necessary a decoder first to work with uncompressed audio?

If is possible, EnsureVBRMP3Sync() must be before AudioDubEx(BlankClip(...)) or after?

Petitions
With buttons 'New' and 'Edit' in Source, DSP and Destination we can test others decoders, DSP functios and encoders.

Please include the Downmix - DPL I, like DPL II but:
ssr = MixAudio(sl, sr, 0.2222, 0.2222)
ssl = Amplify(ssr, -1.0)


Bugs
The last buffer send to the encoder can be incomplete but is send complete, then there are extrabytes at end.
The RiffChunkSize in wav header is incorrect.

A example (without DSP), the same Input.wav and three outputs:
Code:
Input.wav (with correct classical wav header)
--------------
FileLength ..:  1920044 bytes
RiffChunkSize:  1920036     Ok = FileLength - 8 = DataLength + 36
DataLength ..:  1920000
Duration  ...:  10.000000 sec.

Output.wav (RiffChunkSize error, Extrabytes at end of file)
--------------
FileLength ..:  1933356 bytes
RiffChunkSize:  1920450      Error: Must be FileLength - 8 = 1933348
DataLength ..:  1920000      Warning: ExtraBytes at end of file, bytes = 13312
Duration  ...:  10.000000 sec. (with DataLength Ok, extrabytes are ignored)

Output.dat (Raw PCM, without header)
--------------
FileLength ..:  1933312 bytes
Duration  ...:  10.069333 sec.  Error: Extrabytes interpreted as data

Output.ogg
--------------
Duration  ...:  10.069333 sec.  Error: Extrabytes interpreted as data
Another warning: with source 5.1, 48 KHz (ac3, aac, dts, ...) greater than 124 min. 16.54 sec
m_nSizeInBytes = m_nSampleCount * nBlockAlign is > 2^32, then we need a int64 variable in source code.

Last edited by tebasuna51; 14th September 2009 at 01:08. Reason: Deleted link to Shon3i package because donīt wok with actual BeHappy versions
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Old 28th December 2005, 11:04   #3  |  Link
chros
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Isn't there mp3 output yet ?
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Old 28th December 2005, 12:34   #4  |  Link
dimzon
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@tebasuna51
Upmix is not forking yet, this is just algorytm, not working script yet!

EnsureVBRMP3Sync - I think it must be right after decoder. I believe this is just buffering to avoid desync and it can be used with any source type (maybe i'm wrong?)

Petitions
Quote:
Please include the Downmix - DPL I, like DPL II but:
ssr = MixAudio(sl, sr, 0.2222, 0.2222)
ssl = Amplify(ssr, -1.0)
I beleive you can do it yourself - just edit downmix.extension via notepad - I believe you are able to do it. And don't forget to share modified file with me

Quote:
With buttons 'New' and 'Edit' in Source, DSP and Destination we can test others decoders, DSP functios and encoders.
there are some troubles implementing this functionality... But you can add new Source, DSP and Destination manualy:
Creating new Source
Use nicaudio.extension as starting point for new source. Copy it to yorsource.extension and open via notepad. Don't forget to generate new UUID(GUID) for UniqueID attribute
Edit Name attribute and
Edit <Script> element
{0} means input file name
{1} means output file name
{2} means unique string (to use as part of identifier)
{3} means '{' character (to allow '{' to be used)
{4} means '}' character (to allow '}' to be used)
Add/Delete/Edit <SupportedFileExtension> element

Sample:
Code:
<?xml version="1.0"?>
<BeHappy.Extension 
    xmlns:xsd="http://www.w3.org/2001/XMLSchema" 
    xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" 
    xmlns="http://workspaces.gotdotnet.com/behappy">
  <AudioSource Name="MySourceName" UniqueID="5D209A6D-E6EA-4640-AF60-BAE14A529792">
    <Script>SomeSource("{0}")</Script>
    <SupportedFileExtension>ext1</SupportedFileExtension>
    <SupportedFileExtension>ext2</SupportedFileExtension>
    <SupportedFileExtension>ext3</SupportedFileExtension>
  </AudioSource>
</BeHappy.Extension>
Creating new DSP
There are 2 ways to create simple DSP:
Creating very simple DSP
Example:
Code:
<BeHappy.Extension
	xmlns:xsd="http://www.w3.org/2001/XMLSchema"
	xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
	xmlns="http://workspaces.gotdotnet.com/behappy">
  <AudioDSP Name="Normalize to 100%" UniqueID="6158f79f-d8a0-4021-89ae-b77b37c04c55">
    <Script>
# Some script:
Normalize()
    </Script>
  </AudioDSP>
</BeHappy.Extension>
Creating multioption DSP
Take look @ DownMix.extension and UpMix.extension - good examples of multioption DSP
Code:
<?xml version="1.0"?>
<BeHappy.Extension
	xmlns:xsd="http://www.w3.org/2001/XMLSchema"
	xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
	xmlns="http://workspaces.gotdotnet.com/behappy">
  <AudioDSP UniqueID="9579E57B-2D27-4583-99A4-921718E25B41">
    <Plugin>
      <MultiOptionDSP Type="BeHappy.Extensions.MultiOptionDSP, BeHappy">
        <TitleFormatString>DSP Name, Selected Option = {0}</TitleFormatString>
        <ScriptPrologue>
# ScriptPrologue
# this part of script will be included in resulting script without any conditions
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
        </ScriptPrologue>
        <Option>
          <Name>Option Name  1</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  2</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  3</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  4</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  5</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  6</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <Option>
          <Name>Option Name  7</Name>
          <Value>
# this part of script will be included in resulting script only if this option is selected
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
          </Value>
        </Option>
        <ScriptEpilogue>
# ScriptEpilogue
# this part of script will be included in resulting script without any conditions
# {0} means input file name
# {1} means output file name
# {2} means unique string (to use as part of identifier)
# {3} means '{' character (to allow '{' to be used)
# {4} means '}' character (to allow '}' to be used)
        </ScriptEpilogue>
      </MultiOptionDSP>
    </Plugin>
  </AudioDSP>
</BeHappy.Extension>
Creating new Destination
Creating new Destination is very close to new Source. Take look @ ffmpeg-ac3.extension, enc_aacPlus.extension, wavpack.extension, flac.extension
<ExecutableArguments>
{0} means output file name
{1} means samplerate in Hz
{2} means bits per sample
{3} means channel count
{4} means samplecount
{5} means size in bytes

Example:
Code:
<?xml version="1.0"?>
<BeHappy.Extension
	xmlns:xsd="http://www.w3.org/2001/XMLSchema"
	xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
	xmlns="http://workspaces.gotdotnet.com/behappy">
	<AudioEncoder Name="EncoderName" UniqueID="83517DA6-B34F-45ee-B48C-5D9370CC6032">
		<Script>#Some script to be used for this encoder</Script>
		<ExecutableFileName>encoder.exe</ExecutableFileName>
		<ExecutableArguments>- "{0}" --cbr 32000</ExecutableArguments>
		<SupportedFileExtension>ext1</SupportedFileExtension>
		<SupportedFileExtension>ext2</SupportedFileExtension>
		<SupportedFileExtension>ext3</SupportedFileExtension>
	</AudioEncoder>
</BeHappy.Extension>
Creating a really complex Source/DSP/Encoder with it's own GUI
If you are familiar with .NET you can create really fine and complex extension. Take look @ BeHappy.OggVorbis.Encoder.Extension project.
You must:
  • Create Class Library
  • Add reference to BeHappy.Extensibility.dll
  • Impelent IDigitalSignalProcessor or IAudioSource or IAudioEncoder interface
  • Optionally implement ISupportConfiguration interface
  • Create .extension file (refer to BeHappy.OggVorbis.Encoder.extension for example)

PS. One *.extension can contain any amount of DSP Source and Encoder at same time

Last edited by dimzon; 28th December 2005 at 13:44.
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Old 28th December 2005, 14:11   #5  |  Link
dimzon
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fresh beta out

Last edited by dimzon; 23rd May 2006 at 21:22.
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Old 28th December 2005, 15:12   #6  |  Link
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Quote:
Originally Posted by tebasuna51
Bugs
The last buffer send to the encoder can be incomplete but is send complete, then there are extrabytes at end.
Finally I found a bug in Microsoft VfW AVIStreamRead function - it still "read" data even if EOF is alredy occured....

Now I'm using such workaround:
Code:
int nHowMany = Math.Min((int) (m_aviStreamInfo.dwLength-frameSample), MAX_SAMPLES_PER_ONCE) ;
AVIStreamRead(m_aviStream, frameSample, nHowMany, h.AddrOfPinnedObject(), frameBufferTotalSize, out bytesRead, out samplesRead);
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Old 28th December 2005, 19:42   #7  |  Link
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Nero7 AAC encoder extension is avaluable



download it @ workspace @ gotdotnet.com

Last edited by dimzon; 28th December 2005 at 20:00.
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Old 29th December 2005, 02:22   #8  |  Link
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Thanks for your rapid and large answer. I have a lot of work to do.
First I want answer your questions.
Quote:
EnsureVBRMP3Sync - I think it must be right after decoder. I believe this is just buffering to avoid desync and it can be used with any source type (maybe i'm wrong?)
I think is not necessary for uncompressed audio, but I'm not sure.
Quote:
Which software did You use for such diagnostic?
I make a little utility to read wav header. But with a hex editor (I use WinHex) you can see the header (only 44 bytes). Here is a graphical description of wav header:
http://ccrma.stanford.edu/courses/42...ts/WaveFormat/
Quote:
I can use int64 in source BUT there are only 4 bytes in WAV HEADER - how I must write this to WAV?
This is the problem for the wav limit 4 GB. And 2 GB limit is for soft using 'signed long int' for these fields.
The faad decoder (aac) can generate 'invalid' wav's > 4 GB, and fill these fields (RiffLength and DataLength) with 0xFFFFFF00.
Foobar also make 'invalid' wav's > 4 GB, and fill these fields with the 4 low order bytes.
Really the only way, afaik, to use a wav > 4 GB is split it in mono wav's. Then Tranzcode, WaveWizard and BeSplit can generate valid (not for BeSplit because a bug in BlockAlign field) mono wav's with the original length. The method is ignore these fields and continue working until end of file.
I prefer the faad method (0xFFFFFF00) because Foobar method can be confused with valid fields.
Quote:
Can you fix my source?
I send you a PM (2005-12-19) about this. In your previous BeHappy version you use for DataLength:
Code:
Int64 nByteSize = streamSampleLength * m_WavHeader.nBlockAlign;
target.Write(BitConverter.GetBytes((uint)Math.Min(nByteSize,uint.MaxValue)),0,4);
And I suggest you use the same method for RiffLength. In actual version the int64 is missing and you use for DataLength the method from RiffLength (?).

In the actual version, the variable with possible overflow is used also in:
Code:
private uint m_nSizeInBytes;
...
m_nSizeInBytes = m_nSampleCount*m_wavHeader.nBlockAlign;  // may be > 2^32
...
private void createEncoderProcess()
...
  info.Arguments = string.Format(m_commandLine, m_output,
   m_wavHeader.nSamplesPerSec, m_wavHeader.wBitsPerSample,
   m_wavHeader.nChannels,m_nSampleCount,m_nSizeInBytes);
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Old 29th December 2005, 10:56   #9  |  Link
dimzon
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Quote:
Originally Posted by tebasuna51
I prefer the faad method (0xFFFFFF00) because Foobar method can be confused with valid fields.
pleaese check this code:

Code:
private long m_nSizeInBytes;
...	
private void writeHeader(Stream target )
{
	const uint FAAD_MAGIC_VALUE = 0xFFFFFF00;
	const uint WAV_HEADER_SIZE = 36;
	bool useFaadTrick =  m_nSizeInBytes>=(uint.MaxValue-WAV_HEADER_SIZE);
	target.Write(System.Text.Encoding.ASCII.GetBytes("RIFF"),0,4);
	target.Write(BitConverter.GetBytes(useFaadTrick?FAAD_MAGIC_VALUE:(uint)(m_nSizeInBytes + WAV_HEADER_SIZE)),0,4);
	target.Write(System.Text.Encoding.ASCII.GetBytes("WAVEfmt "),0,8);
	target.Write(BitConverter.GetBytes((uint)0x10),0,4);
	target.Write(BitConverter.GetBytes(m_wavHeader.wFormatTag),0,2);
	target.Write(BitConverter.GetBytes(m_wavHeader.nChannels),0,2);
	target.Write(BitConverter.GetBytes(m_wavHeader.nSamplesPerSec),0,4);
	target.Write(BitConverter.GetBytes(m_wavHeader.nAvgBytesPerSec),0,4);
	target.Write(BitConverter.GetBytes(m_wavHeader.nBlockAlign),0,2);
	target.Write(BitConverter.GetBytes(m_wavHeader.wBitsPerSample),0,2);
	target.Write(System.Text.Encoding.ASCII.GetBytes("data"),0,4);
	target.Write(BitConverter.GetBytes(useFaadTrick?FAAD_MAGIC_VALUE:(uint)m_nSizeInBytes),0,4);
}
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Old 29th December 2005, 12:26   #10  |  Link
tebasuna51
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@Dimzon
Yes, I think can work OK.
Thanks.
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Old 29th December 2005, 13:45   #11  |  Link
dimzon
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introducing new feature - Multioption encoder (wait for fresh beta)
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Old 29th December 2005, 14:54   #12  |  Link
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I don't know whether you are already doing this:

any_audiosurce()
ConvertAudioTo24bit() #or better float or 32bit
downmix() / upmix() / normalize()
ConvertAudioTo16bit()

this should effectively reduce rounding errors.
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Old 29th December 2005, 15:07   #13  |  Link
dimzon
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Quote:
Originally Posted by scharfis_brain
I don't know whether you are already doing this:

any_audiosurce()
ConvertAudioTo24bit() #or better float or 32bit
downmix() / upmix() / normalize()
ConvertAudioTo16bit()

this should effectively reduce rounding errors.
thanx, i will implement it nex year!
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Old 29th December 2005, 15:27   #14  |  Link
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New beta is out!
Get it @ workspace @ gotdotnet.com
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Old 29th December 2005, 15:32   #15  |  Link
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Usually, ac3 to ac3 encodings with besweet (and also with ffmpeg as i heard) has a problem, that resulting ac3 has low volume.
Has BeHappy same problem or does it manage this issue well?
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Old 29th December 2005, 15:43   #16  |  Link
dimzon
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Quote:
Originally Posted by Prodater64
Usually, ac3 to ac3 encodings with besweet (and also with ffmpeg as i heard) has a problem, that resulting ac3 has low volume.
Has BeHappy same problem or does it manage this issue well?
BeHappy use ffmpeg for AC3 encoding bcz there are not any another free avaluable AC3 encoder
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Old 29th December 2005, 16:51   #17  |  Link
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Hey! Now we can create more advanced DSP using Sox Audio Effect Filter !
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Old 29th December 2005, 17:43   #18  |  Link
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I need Your help!

2 All
Hi! You can help BeHappy project!
I need:
New DSP scripts - Dinamyc Range Compression e.t.c
BeHappy logo & icons
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Old 30th December 2005, 17:24   #19  |  Link
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Now BeHappy works Ok with wav header and without extrabytes at end.

I make a test with a source 5.1, 48 KHz (ac3) > 132 min. and BeHappy works Ok for wav > 4 GB and for a transcode to aac.

Relevant info of input.ac3
Code:
SampleRate                     0 : 48000 KHz.
BitRate                       15 : 448 Kb/s.
Audio coding mode (acmod)      7 : 3/2 - L, C, R, SL, SR
Low Frequency Effects channel  1 : Present
FileSize      : 444358656 bytes.
Frames        : 247968
Duration      : 7934.976 seconds ( 2 h. 12 m. 14.976 s.)
Relevant info of output.wav
Code:
FileSize ....:  4570546220
RiffLength...:  4294967040  (= 0xFFFFFF00) Error: ... (always with > 4GB)
NumChannels .:  6
SampleRate ..:  48000
OffsetData ..:  44
DataLength ..:  4294967040  (= 0xFFFFFF00) Warning: assumed Datalength = 4570546176 (FileSize - 44)
Duration  ...:  7934.976 sec., (2h. 12m. 14.976 s.)
For aac transcode I use:
behappy.extension.encoder.nero7aac.exe -o "D:\...\132.mp4"
-rr 48000 -rb 16 -rc 6 -vbr_streaming
-aacprofile_he -codecquality_high

Relevant info of output.mp4 (from Foobar2000 v0.8.3)
Code:
bitrate = 191
samplerate = 48000
channels = 6
aac_profile = HE AAC
codec = AAC
tool = Nero AAC Codec 4.2.1.0
samples = 380878848
FileSize = 189,944,318 Bytes (181.15 MB)
Length = 2:12:14.976
As you can see the duration is preserved exactly. Congratulations Dimzon.
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Old 31st December 2005, 04:40   #20  |  Link
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What ideas do you have for the logo? (colors, style, some logos you like, etc.)

To buttress the Hun's suggestion: http://www.digitalprosound.com/Htm/S...ap2_Apogee.htm

It deals with sample rate, not resolution but the concept still applies, expand to at 2x, preferably 4x, do the modifications, downsample.
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