discogs link
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Showing posts with label music. Show all posts
Showing posts with label music. Show all posts
Tuesday, March 20, 2012
Friday, November 11, 2011
#quake ost (1996)
i like the soundtrack of the first quake a lot. used to play the game back in 1996...
never knew it was produced by trent reznor and not a big fan of his other work in general, but this one is really well done.
--
Sunday, July 10, 2011
Tuesday, February 15, 2011
#"free albums galore"
a selective collection of free albums for download.
some nice jazz bits over there.
http://freealbums.blogsome.com/
some nice jazz bits over there.
http://freealbums.blogsome.com/
Monday, February 8, 2010
#project52: or how to write a track a week (for an year)
mr. corey layman (http://developermusic.com) has started a project - "an attempt to record and release a new song every week for one year"
check it out:
http://developermusic.com/blog/
amazing and quite inspiring as well :-)! if you are having a "writers block" or just looking for some inspiration you might want to check this guy out. this basically proves how creative people can be.
great stuff
lubomir
check it out:
http://developermusic.com/blog/
amazing and quite inspiring as well :-)! if you are having a "writers block" or just looking for some inspiration you might want to check this guy out. this basically proves how creative people can be.
great stuff
lubomir
#a hiphop track
trig.com seems to be down so i've started uploading tracks to soundclound.com.
a quick hiphop beat:
These Streets
dj premiere sample + some scratching.
"going back to them roots". heh
lubomir
a quick hiphop beat:
These Streets
dj premiere sample + some scratching.
"going back to them roots". heh
lubomir
Wednesday, December 23, 2009
#Techno tribute mixes
Friday, November 20, 2009
#LPs by Dale Jacobs, Diana Express
I've decided to share 2 of my LP's, since they are out of press and very difficult to find on Ebay, Discogs. The "Coastal Zone" is brand new, while the other one is in a fairly good condition. I've used a process to eliminate clicks and pops, while no artifacts were added. All audio files are 320kbps MP3. Photos of record sleeve artwork are included.
If you own the rights to an album in this list and think that it is not appropriate for it to be posted on the internet, drop me a notification email and i will remove it.
Dale Jacobs - Coastal Zone
Third and last album from the late Vancouver based jazz maestro, Dale Jacobs. Recorded for the New Music Records label in the year of 1989. Features Dale Jacobs on keyboards, Gorg McCrostie on guitars and John Ferraro on drums. The album takes you on a warm and relaxed journey of melodic keys, analogue synthesizer sounds, virtuoso guitars and precision drumming. In most of the tracks you may immediately recognize Dale Jacobs's slick signature sound.
Download
Diana Express - Prayer For Rain
"Prayer for rain" is an album by the bulgarian group Diana Express, that includes the self titled cult song. Recorded in the "Balkanton" recording studios. Arrangement by Mitko Shterev, Keyboards - Mitko Shterev, Tsvetan Banov - percussions, Illiya Angelov - lead vocal and guitars, Zhivko Topalov - Bass, Maxim Goranov - lead guitar.
Download
If you own the rights to an album in this list and think that it is not appropriate for it to be posted on the internet, drop me a notification email and i will remove it.
Dale Jacobs - Coastal Zone
Third and last album from the late Vancouver based jazz maestro, Dale Jacobs. Recorded for the New Music Records label in the year of 1989. Features Dale Jacobs on keyboards, Gorg McCrostie on guitars and John Ferraro on drums. The album takes you on a warm and relaxed journey of melodic keys, analogue synthesizer sounds, virtuoso guitars and precision drumming. In most of the tracks you may immediately recognize Dale Jacobs's slick signature sound.
Download
"Prayer for rain" is an album by the bulgarian group Diana Express, that includes the self titled cult song. Recorded in the "Balkanton" recording studios. Arrangement by Mitko Shterev, Keyboards - Mitko Shterev, Tsvetan Banov - percussions, Illiya Angelov - lead vocal and guitars, Zhivko Topalov - Bass, Maxim Goranov - lead guitar.
Download
Saturday, March 21, 2009
#"Dynamic Range" - Pleasurize Music?
Just recently I've found out about this web page:
http://www.pleasurizemusic.com
This time behind the initiative are serious companies such as:
Tischmeyer Technology
Alogorithmics
SPL
Brainworx
Tomo AudioLabs
The initiative of the "Pleasurize Music Foundation" is proposing an open source VU/RMS meter, which can be used to calculate a "Dynamic Range" value of a audio track - offline or real-time with hosted plugin. The main idea is to standard-ise the use of a specific tool in any studio across the world, a tool that determines what the "Dynamic Range" of the tracks examined is and if these track are unnecessary loud. I strongly support this proposal!
Here is more info I've gathered from personal experience:
"Dynamic Range"
From the website explained as:
...This is the average cumulative difference between peak and loudness (RMS) over a specific period of time (duration of a song or album) and is a whole number value given in decibels. Just the top 20% of the loudness are taken into consideration to ensure that songs with a long intro and an over compressed refrain doesn´t appear with a too high DR value.
For the quick tests I did with the "Offline Meter" they are providing, the returned DR values seem completely adequate. Tracks which I think sound super-loud had DR values of 6 for example. Tracks with normal volumes were around 11-14 DR.
Basically the manual of the tools says that an acoustical musical performance for example should be around DR=12, DR=14. For electronic club music I guess it can go a little bit lower around DR=10. But still! This means that the limiters will begin to kick in and DR=12 still will be absolutely reasonable value.
The "Offline Meter"
The offline meter can be used outside of your DAW application (since it is a standalone exe). Loads and analyzes the tracks very fast! The DR value returned from the meter should be considered the "official" DR value for this track. Not the DR values returned from the "Realtime Meter"
The Plug-in "Real-time Meter"
So far the plugin version of the DR meter is only available in the VST format. They are saying that AU/RTAS versions are on the way too.
The plug-in version loads almost instantaneously on my old PC. It has peak meter, RMS meter, mono switch, link switch, and DR value meter along side a very nice moving colored block which turns from green to red if the DR values begin to lower too much.
The only remark I must make here is that apparently the RMS has "Display gain" value of 3dB applied to it. So if you compare the RMS values with other meters (such as Voxengo Span for example) you will have to add 3db more to DR meter values to make them equal. So far I think they are using a 300ms window for RMS calculation which is ok.
Both the plug-in and the offline versions are developed by Tischmeyer and Alogorithmics. Algorithmics alongside Isotope are probably one of the most respected DSP code algorithm suppliers. Tischmeyer Technology is an offspring of Tischmeyer Publishing GMBH with their excellent mastering studio located in Germany and publishing of mastering books and tutorials.
So far so good. I've already started using the new meter. If you are interested give it a try, register on their website and spread the word on forums and blogs if you support the initiative.
Regards
Lubomir
http://www.pleasurizemusic.com
This time behind the initiative are serious companies such as:
Tischmeyer Technology
Alogorithmics
SPL
Brainworx
Tomo AudioLabs
The initiative of the "Pleasurize Music Foundation" is proposing an open source VU/RMS meter, which can be used to calculate a "Dynamic Range" value of a audio track - offline or real-time with hosted plugin. The main idea is to standard-ise the use of a specific tool in any studio across the world, a tool that determines what the "Dynamic Range" of the tracks examined is and if these track are unnecessary loud. I strongly support this proposal!
Here is more info I've gathered from personal experience:
"Dynamic Range"
From the website explained as:
...This is the average cumulative difference between peak and loudness (RMS) over a specific period of time (duration of a song or album) and is a whole number value given in decibels. Just the top 20% of the loudness are taken into consideration to ensure that songs with a long intro and an over compressed refrain doesn´t appear with a too high DR value.
For the quick tests I did with the "Offline Meter" they are providing, the returned DR values seem completely adequate. Tracks which I think sound super-loud had DR values of 6 for example. Tracks with normal volumes were around 11-14 DR.
Basically the manual of the tools says that an acoustical musical performance for example should be around DR=12, DR=14. For electronic club music I guess it can go a little bit lower around DR=10. But still! This means that the limiters will begin to kick in and DR=12 still will be absolutely reasonable value.
The "Offline Meter"
The offline meter can be used outside of your DAW application (since it is a standalone exe). Loads and analyzes the tracks very fast! The DR value returned from the meter should be considered the "official" DR value for this track. Not the DR values returned from the "Realtime Meter"
The Plug-in "Real-time Meter"
So far the plugin version of the DR meter is only available in the VST format. They are saying that AU/RTAS versions are on the way too.
The plug-in version loads almost instantaneously on my old PC. It has peak meter, RMS meter, mono switch, link switch, and DR value meter along side a very nice moving colored block which turns from green to red if the DR values begin to lower too much.
The only remark I must make here is that apparently the RMS has "Display gain" value of 3dB applied to it. So if you compare the RMS values with other meters (such as Voxengo Span for example) you will have to add 3db more to DR meter values to make them equal. So far I think they are using a 300ms window for RMS calculation which is ok.
Both the plug-in and the offline versions are developed by Tischmeyer and Alogorithmics. Algorithmics alongside Isotope are probably one of the most respected DSP code algorithm suppliers. Tischmeyer Technology is an offspring of Tischmeyer Publishing GMBH with their excellent mastering studio located in Germany and publishing of mastering books and tutorials.
So far so good. I've already started using the new meter. If you are interested give it a try, register on their website and spread the word on forums and blogs if you support the initiative.
Regards
Lubomir
Saturday, February 7, 2009
#Inter-sample peaks and audio compression
Inter-sample peaks have been more of a problem recently with the constant struggle for louder sound. This post includes some tests of the consistency of ISP clear audio signals after performing compression .
The above article at Gearslutz explains inter-sample peaks quite well. Just to simplify it a little bit here is a time-domain plot of a sample. This could be part of any type of sound - bass guitar, synthesizer, but this is more of a 'designed' plot, since the chance of a similar sequence appearing is minimal (there are drastic changes in sample values).
view image
The areas marked with 'ISP' are the actual problems. ISPs are a digital-to-analog (DAC) conversation issue. Your digital medium such as CD-A may contain tracks that does not clip at all on normal meters, but in theory there is a chance that at certain sample configurations such as the shown above, the reconstructed (via interpolation) analog audio signal, exceeds 0 db. Such peaks will be introduced as distortion in listening conditions.
"Would this type of distortion be really that audible?"
It really depends on the playback system (DAC), the listener, the audio material. But it may be quite audible.
"Well how to prevent it?"
There are algorithms in some DSP limiters, that deal with the problem quite well. From my understanding of interpolation in mathematics, the only way to deal with such problems is to change values of the neighbor points (samples) forward and probably backwards in the time domain. Another important thing that must be done is to over-sample the signal so that more points are introduced in the plot.
Here is the list of the know limiters that prevent ISP:
Sonnox Limiter, Voxengo Elephant, TC Limiter, Izotope Ozone, PSP Xenon and others.
Note, that you must engage the oversampling mode of the limiter to enable the desired functionality.
As you may know most of these limiters are commercial products. The only free tools you would find are the meters, such as: John Schwartz's Bitter, SSL's X-ISM and others.
Still, if you are willing to buy a DSP limiter for your needs, I could not recommend a better limiter than Elephant. Its the cheapest, the most flexible and arguably the best sounding one.
Do the listed above tools really work for ISP?
Well here is the tricky part. There are certain conditions in which the ISP prevention may not work for any of the above products. However most of these conditions are quite extreme.
Recently I've conducted some tests with two very popular audio compression formats - MP3 and FLAC. The idea behind the test was a simple question: Does audio compression over a ISP free signal, introduces back the ISPs? Well the answer to that question ended up slightly complicated.
My first test was with a "designed" WAV file with series of samples that would positively introduce ISPs:
view image
I've used Elephant 2 to add 24db of gain to the signal, while oversampling to x2 and limiting at -0.1 db. Then created a MP3 file directly from Reaper with LAME encoder at 320kbps, normal quality. While decoding the resulted MP3 file into Cooledit I've noticed that there is no sample clipping or eventual inter-sample peaks. Then I repeated the same test with lower bitrate MP3 - still no problems were found, after the decoding. But then I've realized that this test is flawed for the following reasons:
- the frequency response of the input signal wasn't really 'sufficient' for testing (it was just a waveform peaking around 2khz)
- the average RMS level after limiting was still quite low.
So I'vee decided to do a little bit more testing on the subject. This time with pink noise and some extreme settings.
Without explaining more I will directly post the test results. There are two tests included (for FLAC, MP3) at different RMS levels.
Test files can be downloaded from here:
mirror1
mirror2
Regarding the MP3 tests:
Decoded MP3 audio signal would only clip if the RMS of the input is very high.
There are ways to deal with this problem:
You can use the --scale command in LAME or tools like mp3gain to fix this.
But still this is sample clipping. There is a chance that no ISP will be present after normalizing the MP3 singal!
Still I should point out that there are way to many parameters involved, when conduction these tests and the test results may vary a lot.
I've also did a another much more extreme test with -1db RMS values (not included). Various limiters did not perform well in prevent ISP in these extreme conditions.
Excessive levels does not do well for ISP prevention and for audio compression.
"How to deal with compression of audio?"
- always use ISP prevention over the track.
- lower the limiter roof to -0.3, -0.5db, or even -1db
- decode the resulted mp3 file with an audio editor and check for clipping
- use statistics in your encoder (such as --clipdetect in LAME)
- of course use the best encoder possible
But still, as the above test proved moderate RMS levels, may not need any normalizations and lowering of the limiter roof at all.
Lubomir
The above article at Gearslutz explains inter-sample peaks quite well. Just to simplify it a little bit here is a time-domain plot of a sample. This could be part of any type of sound - bass guitar, synthesizer, but this is more of a 'designed' plot, since the chance of a similar sequence appearing is minimal (there are drastic changes in sample values).
view image
The areas marked with 'ISP' are the actual problems. ISPs are a digital-to-analog (DAC) conversation issue. Your digital medium such as CD-A may contain tracks that does not clip at all on normal meters, but in theory there is a chance that at certain sample configurations such as the shown above, the reconstructed (via interpolation) analog audio signal, exceeds 0 db. Such peaks will be introduced as distortion in listening conditions.
"Would this type of distortion be really that audible?"
It really depends on the playback system (DAC), the listener, the audio material. But it may be quite audible.
"Well how to prevent it?"
There are algorithms in some DSP limiters, that deal with the problem quite well. From my understanding of interpolation in mathematics, the only way to deal with such problems is to change values of the neighbor points (samples) forward and probably backwards in the time domain. Another important thing that must be done is to over-sample the signal so that more points are introduced in the plot.
Here is the list of the know limiters that prevent ISP:
Sonnox Limiter, Voxengo Elephant, TC Limiter, Izotope Ozone, PSP Xenon and others.
Note, that you must engage the oversampling mode of the limiter to enable the desired functionality.
As you may know most of these limiters are commercial products. The only free tools you would find are the meters, such as: John Schwartz's Bitter, SSL's X-ISM and others.
Still, if you are willing to buy a DSP limiter for your needs, I could not recommend a better limiter than Elephant. Its the cheapest, the most flexible and arguably the best sounding one.
Do the listed above tools really work for ISP?
Well here is the tricky part. There are certain conditions in which the ISP prevention may not work for any of the above products. However most of these conditions are quite extreme.
Recently I've conducted some tests with two very popular audio compression formats - MP3 and FLAC. The idea behind the test was a simple question: Does audio compression over a ISP free signal, introduces back the ISPs? Well the answer to that question ended up slightly complicated.
My first test was with a "designed" WAV file with series of samples that would positively introduce ISPs:
view image
I've used Elephant 2 to add 24db of gain to the signal, while oversampling to x2 and limiting at -0.1 db. Then created a MP3 file directly from Reaper with LAME encoder at 320kbps, normal quality. While decoding the resulted MP3 file into Cooledit I've noticed that there is no sample clipping or eventual inter-sample peaks. Then I repeated the same test with lower bitrate MP3 - still no problems were found, after the decoding. But then I've realized that this test is flawed for the following reasons:
- the frequency response of the input signal wasn't really 'sufficient' for testing (it was just a waveform peaking around 2khz)
- the average RMS level after limiting was still quite low.
So I'vee decided to do a little bit more testing on the subject. This time with pink noise and some extreme settings.
Without explaining more I will directly post the test results. There are two tests included (for FLAC, MP3) at different RMS levels.
input: 20 sec pink noise (WN trought 6pole 'pink' filter, simple pseudo-random
numbers generation, extended with XOR)
limiter: voxengo elephant2, default settings, 4x os, roof -0.2db, test dependant
input gain.
decoding: outputs decodeded with reaper 2.55
meter: SSL X-ISM
### test 1 -12db rms:
output wav (-12db_rms_24bit.wav): 24bit, 44100hz, no dither
* result: no ISP & no clipping
output mp3 (-12db_rms_320kbps.mp3): lame 3.98, CBR 320kbps, normal q
* result: no ISP & no clipping on decode
output mp3 (-12db_rms_96kbps.mp3): lame 3.98, CBR 96kbps, normal q
* result: no ISP & no clipping on decode
output flac (-12db_rms_q8.flac): flac 1.2.1, 24bit, q8
* result: no ISP & no clipping on decode
output flac (-12db_rms_q0.flac): flac 1.2.1, 24bit, q0
* result: no ISP & no clipping on decode
### test 2 -6db rms:
output wav (-6db_rms_24bit.wav): 24bit, 44100hz, no dither
* result: no ISP & no clipping
output mp3 (-6db_rms_320kbps.mp3): lame 3.98, CBR 320kbps, normal q
* result: ISP present & clipping on decode
output mp3 (-6db_rms_96kbps.mp3): lame 3.98, CBR 96kbps, normal q
* result: ISP present & clipping on decode
output flac (-6db_rms_q8.flac): flac 1.2.1, 24bit, q8
* result: ISP present & no clipping on decode
output flac (-6db_rms_q0.flac): flac 1.2.1, 24bit, q0
* result: no ISP & no clipping on decode
### Tests summary:
Lower RMS levels for both MP3 and FLAC decode with no clipping and ISP.
Decoding of mp3 streams with higher RMS level introduce clipping,
but when normalized, ISPs are not present. Higher RMS level flac q8 does
not clip, but strangly ISP are present. Lower compression q0 for FLAC preforms
very well for higher RMS values.
The tested compressors deal much better with lower RMS values.
Test files can be downloaded from here:
mirror1
mirror2
Regarding the MP3 tests:
Decoded MP3 audio signal would only clip if the RMS of the input is very high.
There are ways to deal with this problem:
lame -b 320 --clipdetect [file]
...
WARNING: clipping occurs at the current gain. Set your decoder to decrease
the gain by at least 0.1dB or encode again using --scale [arg]
(For a suggestion on the optimal value of [arg] encode
with --scale 1 first)
You can use the --scale command in LAME or tools like mp3gain to fix this.
But still this is sample clipping. There is a chance that no ISP will be present after normalizing the MP3 singal!
Still I should point out that there are way to many parameters involved, when conduction these tests and the test results may vary a lot.
I've also did a another much more extreme test with -1db RMS values (not included). Various limiters did not perform well in prevent ISP in these extreme conditions.
Excessive levels does not do well for ISP prevention and for audio compression.
"How to deal with compression of audio?"
- always use ISP prevention over the track.
- lower the limiter roof to -0.3, -0.5db, or even -1db
- decode the resulted mp3 file with an audio editor and check for clipping
- use statistics in your encoder (such as --clipdetect in LAME)
- of course use the best encoder possible
But still, as the above test proved moderate RMS levels, may not need any normalizations and lowering of the limiter roof at all.
Lubomir
Saturday, January 17, 2009
#"Turn me up" vs the "LW"
The loudness war takes its victims every year. Lots of CDs are released with tracks limited to the extremes and with very high RMS levels. I've recently bought a couple of electronic music CD's. The music on both CD's is great, however the levels of one of them are pushed to the limits. I end up listening more to the less maximized CD - And I honestly enjoy it much more. The same artist from the maximized CD, was a favorite of mine in the days around 1998. His style hasn't changed much but the levels of his new music are just way too much.
Some thoughts on the 'War'
The 'loudness war' hasn't stopped for many years simply because, it always comes down to the end consumer. And the average consumer's needs for quality are not well defined. If you grab a 'squashed' CD and play to a group of ten people, there is a chance, that none of them will notice the levels and sound quality of the CD. The end consumer's needs are always a priority, but they want it really loud in most cases. Some people are more interested how the record sales will go, that the quality of their music.
Artist with no technical background will always try to compete for the volume of their CD. But what about the engineer's involvement into the process? I think engineers are the ones, who are supposed to explain to the artist, that they are losing quality when the mix is pushed hard. The engineer should always give options to the artist: "Would you like it loud, with tonal balance or would you like it well balanced, not that loud, but without losing quality"? If the engineer, always gives these options to the client maybe the client will start to understand and care more about his own record.
I'm more on the hunt to buy old records which aren't remastered, than taking my chances with modern music that is squashed ('in the face'/'wall of sound') nowadays.
After all, sound quality is not a function of loudness!
'But I like to do everything myself. What levels should I aim for?'
In practice there isn't a single answer to that technical question, simply because the levels you end up with are dependent from the material you are mixing. You shouldn't compare a techno track with a chilled ambient track. The most important bit - learn to distinguish, an unnecessarily loud mix from a well balanced mix. Then try to match the levels of the well balanced mix.
'My track sounds quiet and with less energy compared to other tracks'
Do not fall into the trap. First of all, mix the levels of your track to sound normal! Have you noticed that we use the volume knob of amplifiers less these days. Technically this is not a good thing and speaks bad for the quality of the music we are listening to.
'How do we stop the war?'
As most wars, were started by 'leaders' in the history of human kind. The 'loudness war' should be stopped by the people who originally started it - those who are famous in the music industry. The 'leaders' aka the famous people/groups/bands of POP, METAL, ROCK, HOUSE, R&B music and all other modern styles. Famous mastering engineers should officially declare 'I will not master this record' if the client wants it way too lound. If you get Madonna, Metallica, Benny Benassi, Beyonce and other to release their music with the levels taken into consideration maybe the whole industry will change.
Soon...maybe next year? Not likely, If you ask me. It would be a very slow process to bring back the dynamics. Stop the war now or it will stop the record sales.
'Turn me up'
If you are an artist, a producer, engineer or just a person who appreciates quality music you can support 'Turn me up'.
http://www.turnmeup.org
Famous engineer/producer Charles Dye is behind 'Turn me up' and the idea of the project is simple. First of all its a non commercial project. If you are an artist, who is soon about to release a CD to the market, you can register for a 'Turn me up' certificate and put their logo on your CD. By doing so you can go easy on the levels and not compete with other loud mixes. If you are a mixing/mastering engineer you can offer the ideas of 'Turn me up' to your customers and not use much limiting when working on the tracks.
You can visit their website and read some of the articles on the 'Loudness war' if you are interested.
Lubomir
Some thoughts on the 'War'
The 'loudness war' hasn't stopped for many years simply because, it always comes down to the end consumer. And the average consumer's needs for quality are not well defined. If you grab a 'squashed' CD and play to a group of ten people, there is a chance, that none of them will notice the levels and sound quality of the CD. The end consumer's needs are always a priority, but they want it really loud in most cases. Some people are more interested how the record sales will go, that the quality of their music.
Artist with no technical background will always try to compete for the volume of their CD. But what about the engineer's involvement into the process? I think engineers are the ones, who are supposed to explain to the artist, that they are losing quality when the mix is pushed hard. The engineer should always give options to the artist: "Would you like it loud, with tonal balance or would you like it well balanced, not that loud, but without losing quality"? If the engineer, always gives these options to the client maybe the client will start to understand and care more about his own record.
I'm more on the hunt to buy old records which aren't remastered, than taking my chances with modern music that is squashed ('in the face'/'wall of sound') nowadays.
After all, sound quality is not a function of loudness!
'But I like to do everything myself. What levels should I aim for?'
In practice there isn't a single answer to that technical question, simply because the levels you end up with are dependent from the material you are mixing. You shouldn't compare a techno track with a chilled ambient track. The most important bit - learn to distinguish, an unnecessarily loud mix from a well balanced mix. Then try to match the levels of the well balanced mix.
'My track sounds quiet and with less energy compared to other tracks'
Do not fall into the trap. First of all, mix the levels of your track to sound normal! Have you noticed that we use the volume knob of amplifiers less these days. Technically this is not a good thing and speaks bad for the quality of the music we are listening to.
'How do we stop the war?'
As most wars, were started by 'leaders' in the history of human kind. The 'loudness war' should be stopped by the people who originally started it - those who are famous in the music industry. The 'leaders' aka the famous people/groups/bands of POP, METAL, ROCK, HOUSE, R&B music and all other modern styles. Famous mastering engineers should officially declare 'I will not master this record' if the client wants it way too lound. If you get Madonna, Metallica, Benny Benassi, Beyonce and other to release their music with the levels taken into consideration maybe the whole industry will change.
Soon...maybe next year? Not likely, If you ask me. It would be a very slow process to bring back the dynamics. Stop the war now or it will stop the record sales.
'Turn me up'
If you are an artist, a producer, engineer or just a person who appreciates quality music you can support 'Turn me up'.
http://www.turnmeup.org
Famous engineer/producer Charles Dye is behind 'Turn me up' and the idea of the project is simple. First of all its a non commercial project. If you are an artist, who is soon about to release a CD to the market, you can register for a 'Turn me up' certificate and put their logo on your CD. By doing so you can go easy on the levels and not compete with other loud mixes. If you are a mixing/mastering engineer you can offer the ideas of 'Turn me up' to your customers and not use much limiting when working on the tracks.
You can visit their website and read some of the articles on the 'Loudness war' if you are interested.
Lubomir
Sunday, January 4, 2009
#Reaper beatmatching tutorial
This is a small tutorial in PDF format about beatmatching and mixing tracks in Reaper.
The approach is simple and the results are quite good. I've tried mixing tracks offline with programs like Cooledit, Cubase, Ableton but this method is the fastest so far. You can have a draft CD mix up and running in about 30 minutes. The tutorial also includes some ideas about mixing with EQ & FX envelopes.
View PDF:
Reaper_Tutorial.pdf
Download 22mb (Example project files and mix are included):
Megaupload Link
Sendspace Link
There might be some typos, since I've wrote and compiled the tutorial quite fast.
Lubomir
The approach is simple and the results are quite good. I've tried mixing tracks offline with programs like Cooledit, Cubase, Ableton but this method is the fastest so far. You can have a draft CD mix up and running in about 30 minutes. The tutorial also includes some ideas about mixing with EQ & FX envelopes.
View PDF:
Reaper_Tutorial.pdf
Download 22mb (Example project files and mix are included):
Megaupload Link
Sendspace Link
There might be some typos, since I've wrote and compiled the tutorial quite fast.
Lubomir
#Music at Soundcloud.com
here is my soundclound page with various "attempts" of making music:
http://soundcloud.com/liteon
mostly experimental stuff.
i listen to a lot of jazz and on occasion some classical music, hiphop and electronic but not too much..
never got into rock music (guitars) i'm afraid, but i like a lot of tracks by bands like pantera and ac/dc or anything that sounds good basically.
trig.com is occasionally down but you can check my page there too if its loading ok (has more tracks): http://trig.com/liteon
lubomir
http://soundcloud.com/liteon
mostly experimental stuff.
i listen to a lot of jazz and on occasion some classical music, hiphop and electronic but not too much..
never got into rock music (guitars) i'm afraid, but i like a lot of tracks by bands like pantera and ac/dc or anything that sounds good basically.
trig.com is occasionally down but you can check my page there too if its loading ok (has more tracks): http://trig.com/liteon
lubomir
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