I have briefly tried the "--voice" mode and the "normal" mode when
encoding a purely voice signal (with background noise) at 8kbps, and
have been very impressed with the difference. I would like to compress
the signal more... but 8 is as low as it goes.
The "nomal" mode renders the voice
For my ears, Takehiro's scalefac_scale feature will not give better
results.
The quality of many tracks is a little bit lower. I have found no tracks
with better quality. The file size is reduced, but for my opinion, the
quality is more important. So i think, setting this feature as default
I have briefly tried the "--voice" mode and the "normal" mode when
encoding a purely voice signal (with background noise) at 8kbps, and
have been very impressed with the difference. I would like to compress
the signal more... but 8 is as low as it goes.
The "nomal" mode renders the voice
Hi Gaby!
For my ears, Takehiro's scalefac_scale feature will not give better
results.
The quality of many tracks is a little bit lower. I have found no
tracks
with better quality. The file size is reduced, but for my opinion, the
quality is more important. So i think, setting this
I have briefly tried the "--voice" mode and the "normal" mode when
encoding a purely voice signal (with background noise) at 8kbps, and
have been very impressed with the difference. I would like to compress
the signal more... but 8 is as low as it goes.
The "nomal" mode renders the
Howdy All,
In testing my (comparatively naive) hack of the dist10 encoder, I have
discovered that, while it does OK for music, it has real problems with
speech signals. (Caveat: at our lowest overall bitrate of 300kbps for
combined video/audio, we run the audio at 32kbit mono - though we go
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 04, 2000 4:14 PM
Subject: [MP3 ENCODER] Voice encoding questions
Howdy All,
In testing my (comparatively naive) hack of the dist10 encoder, I have
discovered that, while it does OK for music, it
So my question(s) are: Is the solution to my problem to filter/downsample
(and use joint, when I get around to coding it up)? That seems to be what
is making the difference in the case of LAME; I assume that FhG is using
some filtering as well, though there's no way to disable it and see
1) With my encoder (64kbps stereo CRC), every fricative is almost painful to
listen to, as the pink noise bursts end up being narrow band filtered (due
to lack of bits - only the MDCT coeffs closest to the pole are making it
into the bitstream), and there are occasional weird high frequency
Howdy All,
Thanks for the quick replies!
Gabriel Bouvigne wrote:
If you want to encode voice signals, I'd suggest you to use --voice
or --preset voice
Actually, I want to encode general signals (mostly TV and movies), many of
which have significant voice components, and, unfortunately, many
I've encoded some wavefiles with Exact Audio Copy and Audiograbber using
Lame_enc.dll (3.85). I measured the time needed for encoding at 128 CBR (joint)
stereo. The measured times for encoding at HQ were the same as for encoding
at LQ.
Furthermore, the measured times Exact Audio Copy needed
I like to think that I have fixed at least a few. Now that I've finished
a
first pass clean, rewrite, overhaul, and verify, I'm taking a closer look
at
algorithmic (as opposed to purely implementational) problems, starting
with
the main loop, and probably ending with the #^@% psych model.
Hello alex,
abcc I feel guilty using a list mainly devoted to an open source codec (LAME) to
abcc further the development of ClearBand's 'proprietary' codec. (Is a standards
abcc based codec implementation proprietary? We don't sell the codec - we sell a
abcc multicast system, mostly to ISPs
Try doing a BINARY comparation to see
- Original Message -
From: "Wim" [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 04, 2000 8:25 PM
Subject: [MP3 ENCODER] HQ/LQ switches in lame_enc.dll
I've encoded some wavefiles with Exact Audio Copy and Audiograbber using
We should support an option (-ma for Mode Auto) which switches between -a -mm
for highly correlated channels (r 0.98 = mono), -mj for a normal
correlated signals (r = -1.00...-0.20, 0.20...0.91 = stereo) and -ms for nearly not
correlated signals (dual channel audio with independent audio, i.e.
:: -32000Hz files, filled with noise and
:: "Lame -q1 -d -m j -V 2 -B 192 --lowpass 12.0 Hobbit.wav
:: Hobbit-q1.mp3"
:: and
::
:: "Lame -h -d -v -q1 jo3.wav jo3q1.wav"
::
:: |56 - 3 - 1,8%
:: |||64 - 11 - 6,7%
:: 80 - 52 - 31,7%
:: ||
:: I am looking for a program to check the integrity of mp3's. I know that Nero
:Burning Rom does some sort of check on mp3's before burning them, but I haven't been
:able to find a similar utility to check my MP3's without Nero.
::
:: I thought someone out there should know something
::
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