I want to encode speech. Which coded should i use? OGG or MP3? What quality setting (ogg) and bitrate (mp3) do you think i should use?
Thank you
I think you should try Ogg Vorbis, and the aoTuV beta 4 version. I'm not sure about the quality, but you should try it yourself and find out the quality level that suits you.
i tried the original ogg drop xp. I am extremely satisfied with q -1,00 (nominal bitrate 45kbps). Can it go even lower?
OggDropXPd based on aoTuV beta 4 has a -q-2 mode (~32 kbps).
You can also reduce the samplerate to lower the bitrate: according to some people, it also increase the quality (aoTuV at 32 kbps and 44100 Hz is not really pleasant IMO).
OggDropXPd based on aoTuV beta 4 has a -q-2 mode (~32 kbps).
You can also reduce the samplerate to lower the bitrate: according to some people, it also increase the quality (aoTuV at 32 kbps and 44100 Hz is not really pleasant IMO).
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ok tell me more. I dont have time to test, what sample rate should i use?
i am interested in gettin the smallest size with a decent quality for speech/voice. I have 3 formats available MP3/OGG/WMA. Don't have time to test, i will trust whatever you audio experts recommend me.
Thank you all for your ultra fast replies.
I can't help you. I'm only repeating what some other people said about vorbis and ultra-low bitrate encoding. Sorry. But at such low bitrate I would say that's it's not very hard nor very long to test it on your side
I can't help you. I'm only repeating what some other people said about vorbis and ultra-low bitrate encoding. Sorry. But at such low bitrate I would say that's it's not very hard nor very long to test it on your side
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that would do! what sample rate do you recommend for 43kbps ogg?
As I already said it, I haven't tested it. Therefore, it's very hard for me to recommand you anything (apart testing by yourself).
Since the bloke is desperate: A wireline phone has 4 kHz bandwidth. So sampling rate at 8 kHz should be enough to understand the voice. You might want to choose a bit higher. I do not know what choices are provided.
Why not use speex? http://speex.org/ (http://speex.org/)
It is designed for voice encoding.
16 kHz, 16 bit, mono are perfectly OK for pure speech.
Why not use speex? http://speex.org/ (http://speex.org/)
It is designed for voice encoding.
Yes and it has three modes including wide-band (16 kHz) and ultra-wideband (32kHz)
Originally posted in this thread (http://www.hydrogenaudio.org/forums/index.php?showtopic=35214).
Apart from encoding to mono, can anyone suggest space saving parameters for speech only files?
I recently did a project similar to this. It took a little trial and error, but the settings I ended up with seem to be the best for me.
I started with 16-bit mono WAV files at 22050Hz and converted them to MP3 using LAME 3.96.1 with the following parameters:
-V3 --vbr-new --lowpass 8
These settings create an MP3 file with a bit-rate around 48kbps and an 8kHz low-pass filter, which seems fine for speech.
A typical 45 minute speech will reduce from ~115M (WAV) to ~15M (MP3) in about 35 seconds on my computer (P4 2.8GHz, 1G RAM, Windows XP).
Hope this helps...
I would have looked into Speex or aoTuVb4, maybe also HE-AAC before going mp3 on this. But if it's to be played by the regular DAP's, DVD players or such that only do mp3 and wma, there's been several threads on the forum on best settings for audiobooks and such.
[span style='font-size:8pt;line-height:100%']EDIT: Typo.[/span]
the reason i'm am not going to other codec is that they are not supported by my mp3 player. Unforunatelly, so low bitrates of ogg also are not supported. I played a bit with mp3 and i found that 24 kbps 11kHz are enough for my needs. They are a bit bigger as files than ogg but i guess i'll have to live with than
Thank you all for your fast replies.
Here are some very impressive (in my opinion) mp3 settings for voice:
--abr 16 -a --resample 11 --lowpass 5 --athtype 2 -X3
--alt-preset 24 -a --resample 22 --lowpass 7
The second line is obviously better than the first, but the first line has very small file sizes.
Here are some very impressive (in my opinion) mp3 settings for voice:
--abr 16 -a --resample 11 --lowpass 5 --athtype 2 -X3
--alt-preset 24 -a --resample 22 --lowpass 7
The second line is obviously better than the first, but the first line has very small file sizes.
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i get this error from the
second one
Command: C:\Program Files\Music\MP3\Lame 3.90.3\lame.exe --alt-preset 24 -a --resample 22 --lowpass 7 "C:\Documents " "C:\Documents "
LAME version 3.90.3 MMX (http://www.mp3dev.org/)
-- Compiled at [a href="http://www.hydrogenaudio.org]http://www.hydrogenaudio.org[/url]
-- Check this website for up to date information on the --alt-presets
Error: The bitrate specified is out of the valid range for this preset
When using this mode you must enter a value between "80" and "320"
For further information try: "C:\Program Files\Music\MP3\Lame 3.90.3\lame.exe --alt-preset help"
RazorLame encountered an unknown message from LAME while trying to encode "C:\Documents "!
Encoded 0 files in 0:00:00
There was an unexpected LAME message for one file, please check log for error messages.
I used 16kbps and 11kHz and i get an output of 294. Same file is 360 with the first switch
Here are the full parameter lines I am using in foobars cli
--abr 16 -a --resample 11 --lowpass 5 --athtype 2 -X3 - %d
--alt-preset 24 -a --resample 22 --lowpass 7 - %d
and am using them with lame 3.97a11
I just tried them and they worked perfectly. Must be because you are using 3.90.3 lame.
Wouldn't HE-AAC offer no advantage over LC-AAC because speech has no frequencies high enough to benefit from the SBR? or did i miss something here?