mr.karmalicious
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Glad to see that you found what bitrates the q levels are, since I was about to post that. That should help out your cause a little bit.
An old man finally succumbs to convenience
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Bob A (SD)
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Well I suppose this could be considered overkill but I took EAC and with the latest flavors of OGG and LAME ripped a 54.78Mb Jazz Crusaders cut with a wide variety of options. I also had a 96kps WMA version of this track I ripped earlier.
I compared each version for sound quality as well as visually examining the files on spectrum analyzer display. I was rather surprised to see less high frequency content in the OGG files when compared to the LAME MP3 versions of like bit rate. It appears the lowpass filter is much lower with OGG than with LAME resulting in comparatively truncated high frequencies. I also noted that the OGG files were actually bigger than the LAME files with like parameters above 96k; below they were indeed smaller. I had thought that the OGG files would always be the smallest
. Curious I did some googling and found HydrogenAudio forums where some other folks with limited capacity flash players discussed the same issues I have before me. Many opted for LAME MP3 over OGG. And frankly that's what I'll be doing using the LAME -ap command line with a minimum of 112 bit rate. That's turned out to be the minimally acceptable threshold for me in terms of quality and file size.... at least for now. Again I'm rather surprised at OGG.... unless I'm missing something major here which is entirely possible given my "newbie" status with regard to ripping RedBook CDs with different compression encoders.
--Bob
I compared each version for sound quality as well as visually examining the files on spectrum analyzer display. I was rather surprised to see less high frequency content in the OGG files when compared to the LAME MP3 versions of like bit rate. It appears the lowpass filter is much lower with OGG than with LAME resulting in comparatively truncated high frequencies. I also noted that the OGG files were actually bigger than the LAME files with like parameters above 96k; below they were indeed smaller. I had thought that the OGG files would always be the smallest
--Bob
Bob A (SD)
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I finally found the OGG command line options which I'm posting here to share. The possible solution to the problem I found above with the loss of high frequency content is to use the advanced options to set the lowpass filter at say 21.5kHz as recomended here http://jult.nl/printscreens/aotuv?full=1
Here are the command line options (found at http://linuxcommand.org/man_pages/oggenc1.html
OPTIONS
-h, --help
Show command help.
-r, --raw
Assume input data is raw little-endian audio data with no header
information. If other options are not specified, defaults to
44.1kHz stereo 16 bit. See next three options for how to change
this.
-B n, --raw-bits=n
Sets raw mode input sample size in bits. Default is 16.
-C n, --raw-chan=n
Sets raw mode input number of channels. Default is 2.
-R n, --raw-rate=n
Sets raw mode input samplerate. Default is 44100.
--raw-endianness n
Sets raw mode endianness to big endian (1) or little endian (0).
Default is little endian.
-Q, --quiet
Quiet mode. No messages are displayed.
-b n, --bitrate=n
Sets encoding to the bitrate closest to n (in kb/s).
-m n, --min-bitrate=n
Sets minimum bitrate to n (in kb/s).
-M n, --max-bitrate=n
Sets maximum bitrate to n (in kb/s).
--managed
Set bitrate management mode. This turns off the normal VBR
encoding, but allows hard or soft bitrate constraints to be
enforced by the encoder. This mode is much slower, and may also
be lower quality. It is primarily useful for creating files for
streaming.
-q n, --quality=n
Sets encoding quality to n, between -1 (low) and 10 (high). This
is the default mode of operation, with a default quality level
of 3. Fractional quality levels such as 2.5 are permitted. Nor-
mal quality range is 0 - 10.
--resample n
Resample input to the given sample rate (in Hz) before encoding.
Primarily useful for downsampling for lower-bitrate encoding.
--downmix
Downmix input from stereo to mono (has no effect on non-stereo
streams). Useful for lower-bitrate encoding.
--advanced-encode-option optionname=value
Sets an advanced option. See the Advanced Options section for
details.
-s, --serial
Forces a specific serial number in the output stream. This is
primarily useful for testing.
--discard-comments
Prevents comments in FLAC and Ogg FLAC files from being copied
to the output Ogg Vorbis file.
-o output_file, --output=output_file
Write the Ogg Vorbis stream to output_file (only valid if a sin-
gle input file is specified)
-n pattern, --names=pattern
Produce filenames as this string, with %g, %a, %l, %n, %t, %d
replaced by genre, artist, album, track number, title, and date,
respectively (see below for specifying these). Also, %% gives a
literal %.
-c comment, --comment comment
Add the string comment as an extra comment. This may be used
multiple times, and all instances will be added to each of the
input files specified. The argument should be in the form
"tag=value".
-a artist, --artist artist
Set the artist comment field in the comments to artist.
-G genre, --genre genre
Set the genre comment field in the comments to genre.
-d date, --date date
Sets the date comment field to the given value. This should be
the date of recording.
-N n, --tracknum n
Sets the track number comment field to the given value.
-t title, --title title
Set the track title comment field to title.
-l album, --album album
Set the album comment field to album.
Note that the -a, -t, and -l options can be given multiple times. They
will be applied, one to each file, in the order given. If there are
fewer album, title, or artist comments given than there are input
files, oggenc will reuse the final one for the remaining files, and
issue a warning in the case of repeated titles.
ADVANCED ENCODER OPTIONS
Oggenc allows you to set a number of advanced encoder options using the
--advanced-encode-option option. These are intended for very advanced
users only, and should be approached with caution. They may signifi-
cantly degrade audio quality if misused. Not all these options are cur-
rently documented.
bitrate_average_window=NN
Set the managed bitrate window to NN seconds. The bitrate will
be forced to the specified average over a floating window of
this length. May be fractional (e.g. 3.5)
lowpass_frequency=NN
Set the lowpass frequency to NN kHz.
Here are the command line options (found at http://linuxcommand.org/man_pages/oggenc1.html
OPTIONS
-h, --help
Show command help.
-r, --raw
Assume input data is raw little-endian audio data with no header
information. If other options are not specified, defaults to
44.1kHz stereo 16 bit. See next three options for how to change
this.
-B n, --raw-bits=n
Sets raw mode input sample size in bits. Default is 16.
-C n, --raw-chan=n
Sets raw mode input number of channels. Default is 2.
-R n, --raw-rate=n
Sets raw mode input samplerate. Default is 44100.
--raw-endianness n
Sets raw mode endianness to big endian (1) or little endian (0).
Default is little endian.
-Q, --quiet
Quiet mode. No messages are displayed.
-b n, --bitrate=n
Sets encoding to the bitrate closest to n (in kb/s).
-m n, --min-bitrate=n
Sets minimum bitrate to n (in kb/s).
-M n, --max-bitrate=n
Sets maximum bitrate to n (in kb/s).
--managed
Set bitrate management mode. This turns off the normal VBR
encoding, but allows hard or soft bitrate constraints to be
enforced by the encoder. This mode is much slower, and may also
be lower quality. It is primarily useful for creating files for
streaming.
-q n, --quality=n
Sets encoding quality to n, between -1 (low) and 10 (high). This
is the default mode of operation, with a default quality level
of 3. Fractional quality levels such as 2.5 are permitted. Nor-
mal quality range is 0 - 10.
--resample n
Resample input to the given sample rate (in Hz) before encoding.
Primarily useful for downsampling for lower-bitrate encoding.
--downmix
Downmix input from stereo to mono (has no effect on non-stereo
streams). Useful for lower-bitrate encoding.
--advanced-encode-option optionname=value
Sets an advanced option. See the Advanced Options section for
details.
-s, --serial
Forces a specific serial number in the output stream. This is
primarily useful for testing.
--discard-comments
Prevents comments in FLAC and Ogg FLAC files from being copied
to the output Ogg Vorbis file.
-o output_file, --output=output_file
Write the Ogg Vorbis stream to output_file (only valid if a sin-
gle input file is specified)
-n pattern, --names=pattern
Produce filenames as this string, with %g, %a, %l, %n, %t, %d
replaced by genre, artist, album, track number, title, and date,
respectively (see below for specifying these). Also, %% gives a
literal %.
-c comment, --comment comment
Add the string comment as an extra comment. This may be used
multiple times, and all instances will be added to each of the
input files specified. The argument should be in the form
"tag=value".
-a artist, --artist artist
Set the artist comment field in the comments to artist.
-G genre, --genre genre
Set the genre comment field in the comments to genre.
-d date, --date date
Sets the date comment field to the given value. This should be
the date of recording.
-N n, --tracknum n
Sets the track number comment field to the given value.
-t title, --title title
Set the track title comment field to title.
-l album, --album album
Set the album comment field to album.
Note that the -a, -t, and -l options can be given multiple times. They
will be applied, one to each file, in the order given. If there are
fewer album, title, or artist comments given than there are input
files, oggenc will reuse the final one for the remaining files, and
issue a warning in the case of repeated titles.
ADVANCED ENCODER OPTIONS
Oggenc allows you to set a number of advanced encoder options using the
--advanced-encode-option option. These are intended for very advanced
users only, and should be approached with caution. They may signifi-
cantly degrade audio quality if misused. Not all these options are cur-
rently documented.
bitrate_average_window=NN
Set the managed bitrate window to NN seconds. The bitrate will
be forced to the specified average over a floating window of
this length. May be fractional (e.g. 3.5)
lowpass_frequency=NN
Set the lowpass frequency to NN kHz.
bigshot
Headphoneus Supremus
God! I'm glad I'm using a Mac and iTunes!
See ya
Steve
See ya
Steve
Bob A (SD)
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BINGO
The high freqs are back! And ironically the lower q settings with the higher lowpass frequency of 21.5kHz sound comparable to a normal q setting 1 or 2 higher. The major improvements are with the lower q settings. You can see I tried Q1, 2, 3, 4, 4.5, and 5 here. I'm back to OGG
I apologize if these posts annoy or bore anyone. I'm simply sharing my adventures with ripping effectively to a flashplayer.
--Bob
The high freqs are back! And ironically the lower q settings with the higher lowpass frequency of 21.5kHz sound comparable to a normal q setting 1 or 2 higher. The major improvements are with the lower q settings. You can see I tried Q1, 2, 3, 4, 4.5, and 5 here. I'm back to OGG
I apologize if these posts annoy or bore anyone. I'm simply sharing my adventures with ripping effectively to a flashplayer.
--Bob
mr.karmalicious
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Nah, you're not annoying anybody. Congrats on getting EAC with OggEnc up and running, and a HUGE thanks for the command line options!
Bob A (SD)
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Quote:
Thanks. I know you rip at a higher quality setting with OGG but you might find these stats taken from WinAMP file playback interesting.
LAME @ 96kbps settings defaults to a sampling rate of 32KHz
OGG @ Q1 gives an average kbps of 87 sampled at 44KHz
OGG @ Q2 gives an average of 94 kbps at 44KHz
WMA @ 96kbps has a sampling rate of 44KHz
OGG @ Q2 with a lowpass of 21.5KHz gives 100kbps avg at 44KHz
LAME @ 112kbps now defaults to a sampling rate of 44KHz
OGG @ Q3 gives an average kbps of 116 at 44MHz
OGG @ Q3 with a lowpass of 21.5Khz gives an average 119kbps at 44KHz
LAME @ 128kbps sampled at 44KHz
OGG @ Q4 gives an average kbps of 131 at 44KHz
OGG @ Q4 with lowpass give 132 at 44
OGG @ Q4.5 with lowpass gives 144 at 44
LAME @ 160kbps and 44KHz
OGG @ Q5 with or without lowpass gives 171kbps at 44 KHz
Without questioning the effacy of the 96kbps "standard", one can see the advantages of OGG with the lowpass option in giving the best size, kbps and sampling rate when compared with WMA and LAME. Increasing the quality "n" decreases / eliminates the advantages in file size although the avg bit rate still exceeds the reference (e.g Q5=160 but actually gives an average of 171).
With much of what I've recently read indicating that the optimal kbps rate before diminsihing returns applies is between 128 and 196, I can readily understand why OGG Q5 is the recurring highly recommended setting for HD digital players where space isn't a prime issue. From my knothole it would seem to follow that for flashdrive players where size is an issue, OGG Q2 or 3 with the lowpass option invoked or Q4 with or without are very viable and better choices than LAME or WMA. Of course the bottom line is in the listening.
--Bob
| Originally Posted by mr.karmalicious Nah, you're not annoying anybody. Congrats on getting EAC with OggEnc up and running, and a HUGE thanks for the command line options! |
Thanks. I know you rip at a higher quality setting with OGG but you might find these stats taken from WinAMP file playback interesting.
LAME @ 96kbps settings defaults to a sampling rate of 32KHz
OGG @ Q1 gives an average kbps of 87 sampled at 44KHz
OGG @ Q2 gives an average of 94 kbps at 44KHz
WMA @ 96kbps has a sampling rate of 44KHz
OGG @ Q2 with a lowpass of 21.5KHz gives 100kbps avg at 44KHz
LAME @ 112kbps now defaults to a sampling rate of 44KHz
OGG @ Q3 gives an average kbps of 116 at 44MHz
OGG @ Q3 with a lowpass of 21.5Khz gives an average 119kbps at 44KHz
LAME @ 128kbps sampled at 44KHz
OGG @ Q4 gives an average kbps of 131 at 44KHz
OGG @ Q4 with lowpass give 132 at 44
OGG @ Q4.5 with lowpass gives 144 at 44
LAME @ 160kbps and 44KHz
OGG @ Q5 with or without lowpass gives 171kbps at 44 KHz
Without questioning the effacy of the 96kbps "standard", one can see the advantages of OGG with the lowpass option in giving the best size, kbps and sampling rate when compared with WMA and LAME. Increasing the quality "n" decreases / eliminates the advantages in file size although the avg bit rate still exceeds the reference (e.g Q5=160 but actually gives an average of 171).
With much of what I've recently read indicating that the optimal kbps rate before diminsihing returns applies is between 128 and 196, I can readily understand why OGG Q5 is the recurring highly recommended setting for HD digital players where space isn't a prime issue. From my knothole it would seem to follow that for flashdrive players where size is an issue, OGG Q2 or 3 with the lowpass option invoked or Q4 with or without are very viable and better choices than LAME or WMA. Of course the bottom line is in the listening.
--Bob
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