version 1, including all changes.
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perry |
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SoX |
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!!!SoX |
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NAME |
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CONVERSIONS |
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EFFECTS |
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SEE ALSO |
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AUTHOR |
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---- |
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!!NAME |
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soxexam - SoX Examples (CHEAT SHEET) |
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!!CONVERSIONS |
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__Introduction__ |
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In general, SoX will attempt to take an input sound file |
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format and convert it into a new file format using a similar |
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data type and sample rate. For instance, |
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If an output format doesn't support the same data type as |
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the input file then SoX will generally select a default data |
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type to save it in. You can override the default data type |
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selection by using command line options. This is also useful |
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for producing an output file with higher or lower precision |
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data and/or sample rate. |
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Most file formats that contain headers can automatically be |
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read in. When working with header-less file formats then a |
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user must manually tell SoX the data type and sample rate |
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using command line options. |
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When working with header-less files (raw files), you may |
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take advantage of the pseudo-file types of .ub, .uw, .sb, |
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.sw, .ul, and .sl. By using these extensions on your |
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filenames you will not have to specify the corresponding |
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options on the command line. |
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__Precision__ |
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The following data types and formats can be represented by |
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their total uncompressed bit precision. When converting from |
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one data type to another care must be taken to insure it has |
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an equal or greater precision. If not then the audio quality |
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will be degraded. This is not always a bad thing when your |
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working with things such as voice audio and are concerned |
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about disk space or bandwidth of the audio |
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data. |
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Data Format Precision |
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___________ _________ |
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unsigned byte 8-bit |
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signed byte 8-bit |
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u-law 14-bit |
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A-law 13-bit |
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unsigned word 16-bit |
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signed word 16-bit |
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ADPCM 16-bit |
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GSM 16-bit |
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unsigned long 32-bit |
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signed long 32-bit |
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___________ _________ |
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__Examples__ |
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Use the '-V' option on all your command lines. It makes SoX |
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print out its idea of what is going on. '-V' is your |
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friend. |
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To convert from unsigned bytes at 8000 Hz to signed words at |
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8000 Hz: |
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sox -r 8000 -c 1 filename.ub newfile.sw |
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To convert from Apple's AIFF format to Microsoft's WAV |
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format: |
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sox filename.aiff filename.wav |
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To convert from mono raw 8000 Hz 8-bit unsigned PCM data to |
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a WAV file: |
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sox -r 8000 -u -b -c 1 filename.raw |
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filename.wav |
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SoX may even be used to convert sample rates. Downconverting |
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will reduce the bandwidth of a sample, but will reduce |
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storage space on your disk. All such conversions are lossy |
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and will introduce some noise. You should really pass your |
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sample through a low pass filter prior to downconverting as |
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this will prevent alias signals (which would sound like |
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additional noise). For example to convert from a sample |
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recorded at 11025 Hz to a u-law file at 8000 Hz sample |
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rate: |
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sox infile.wav -t au -r 8000 -U -b -c 1 |
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outputfile.au |
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To add a low-pass filter (note use of stdout for output of |
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the first stage and stdin for input on the second |
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stage): |
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sox infile.wav -t raw -s -w -c 1 - lowpass 3700 | sox -t raw |
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-r 11025 -s -w -c 1 - -t au -r 8000 -U -b -c 1 |
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ofile.au |
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If you hear some clicks and pops when converting to u-law or |
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A-law, reduce the output level slightly, for example this |
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will decrease it by 20%: |
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sox infile.wav -t au -r 8000 -U -b -c 1 -v .8 |
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outputfile.au |
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''SoX'' is great to use along with other command line |
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programs by passing data between the programs using |
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pipelines. The most common example is to use mpg123 to |
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convert mp3 files in to wav files. The following command |
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line will do this: |
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mpg123 -b 10000 -s filename.mp3 | sox -t raw -r 44100 -s -w |
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-c 2 - filename.wav |
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When working with totally unknown audio data then the |
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sox -V -t auto filename.snd filename.wav |
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It is important to understand how the internals of |
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''SoX'' work with compressed audio including u-law, |
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A-law, ADPCM, or GSM. ''SoX'' takes ALL input data types |
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and converts them to uncompressed 32-bit signed data. It |
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will then convert this internal version into the requested |
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output format. This means additional noise can be introduced |
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from decompressing data and then recompressing. If applying |
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multiple effects to audio data, it is best to save the |
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intermediate data as PCM data. After the final effect is |
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performed, then you can specify it as a compressed output |
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format. This will keep noise introduction to a |
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minimum. |
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The following example applies various effects to an 8000 Hz |
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ADPCM input file and then end up with the final file as |
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44100 Hz ADPCM. |
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sox firstfile.wav -r 44100 -s -w secondfile.wav |
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sox secondfile.wav thirdfile.wav swap |
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sox thirdfile.wav -a -b finalfile.wav mask |
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Under a DOS shell, you can convert several audio files to an |
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new output format using something similar to the following |
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command line: |
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FOR %X IN (*.RAW) DO sox -r 11025 -w -s -t raw $X |
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$X.wav |
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!!EFFECTS |
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Special thanks goes to Juergen Mueller |
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(jmeuller@uia.au.ac.be) for this write up on |
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effects. |
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__Introduction:__ |
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The core problem is that you need some experience in using |
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effects in order to say |
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Here are some examples which can be used with any music |
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sample. (For a sample where only a single instrument is |
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playing, extreme parameter setting may make well-known |
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Single effects will be explained and some given parameter |
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settings that can be used to understand the theory by |
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listening to the sound file with the added |
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effect. |
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Using multiple effects in parallel or in series can result |
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either in a very nice sound or (mostly) in a dramatic |
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overloading in variations of sounds such that your ear may |
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follow the sound but you will feel unsatisfied. Hence, for |
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the first time using effects try to compose them as |
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minimally as possible. We don't regard the composition of |
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effects in the examples because too many combinations are |
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possible and you really need a very fast machine and a lot |
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of memory to play them in real-time. |
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However, real-time playing of sounds will greatly speed up |
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learning and/or tuning the parameter settings for your |
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sounds in order to get that |
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Basically, we will use the |
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For easy listening of file.xxx ( |
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play file.xxx effect-name effect-parameters |
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Or more SoX-like (for |
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sox file.xxx -t ossdsp -w -s /dev/dsp effect-name |
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effect-parameters |
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or (for |
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sox file.xxx -t sunau -w -s /dev/audio effect-name |
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effect-parameters |
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And for date freaks: |
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sox file.xxx file.yyy effect-name |
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effect-parameters |
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Additional options can be used. However, in this case, for |
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real-time playing you'll need a very fast |
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machine. |
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Notes: |
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I played all examples in real-time on a Pentium 100 with 32 |
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MB and Linux 2.0.30 using a self-recorded sample ( 3:15 min |
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long in |
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Effects: |
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__Echo__ |
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An echo effect can be naturally found in the mountains, |
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standing somewhere on a mountain and shouting a single word |
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will result in one or more repetitions of the word (if not, |
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turn a bit around and try again, or climb to the next |
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mountain). |
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However, the time difference between shouting and repeating |
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is the delay (time), its loudness is the decay. Multiple |
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echos can have different delays and decays. |
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It is very popular to use echos to play an instrument with |
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itself together, like some guitar players (Brain May from |
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Queen) or vocalists are doing. For music samples of more |
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than one instrument, echo can be used to add a second sample |
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shortly after the original one. |
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This will sound as if you are doubling the number of |
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instruments playing in the same sample: |
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play file.xxx echo 0.8 0.88 60.0 0.4 |
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If the delay is very short, then it sound like a (metallic) |
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robot playing music: |
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play file.xxx echo 0.8 0.88 6.0 0.4 |
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Longer delay will sound like an open air concert in the |
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mountains: |
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play file.xxx echo 0.8 0.9 1000.0 0.3 |
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One mountain more, and: |
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play file.xxx echo 0.8 0.9 1000.0 0.3 1800.0 |
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0.25 |
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__Echos__ |
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Like the echo effect, echos stand for |
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The sample will be bounced twice in symmetric |
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echos: |
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play file.xxx echos 0.8 0.7 700.0 0.25 700.0 |
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0.3 |
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The sample will be bounced twice in asymmetric |
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echos: |
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play file.xxx echos 0.8 0.7 700.0 0.25 900.0 |
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0.3 |
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The sample will sound as if played in a garage: |
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play file.xxx echos 0.8 0.7 40.0 0.25 63.0 0.3 |
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__Chorus__ |
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The chorus effect has its name because it will often be used |
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to make a single vocal sound like a chorus. But it can be |
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applied to other instrument samples too. |
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It works like the echo effect with a short delay, but the |
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delay isn't constant. The delay is varied using a sinusoidal |
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or triangular modulation. The modulation depth defines the |
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range the modulated delay is played before or after the |
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delay. Hence the delayed sound will sound slower or faster, |
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that is the delayed sound tuned around the original one, |
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like in a chorus where some vocals are a bit out of |
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tune. |
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The typical delay is around 40ms to 60ms, the speed of the |
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modulation is best near 0.25Hz and the modulation depth |
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around 2ms. |
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A single delay will make the sample more |
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overloaded: |
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play file.xxx chorus 0.7 0.9 55.0 0.4 0.25 2.0 |
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-t |
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Two delays of the original samples sound like |
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this: |
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play file.xxx chorus 0.6 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 |
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0.4 1.3 -s |
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A big chorus of the sample is (three additional |
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samples): |
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play file.xxx chorus 0.5 0.9 50.0 0.4 0.25 2.0 -t 60.0 0.32 |
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0.4 2.3 -t 40.0 0.3 0.3 1.3 -s |
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__Flanger__ |
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The flanger effect is like the chorus effect, but the delay |
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varies between 0ms and maximal 5ms. It sound like wind |
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blowing, sometimes faster or slower including changes of the |
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speed. |
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The flanger effect is widely used in funk and soul music, |
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where the guitar sound varies frequently slow or a bit |
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faster. |
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The typical delay is around 3ms to 5ms, the speed of the |
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modulation is best near 0.5Hz. |
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Now, let's groove the sample: |
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play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -s |
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listen carefully between the difference of sinusoidal and |
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triangular modulation: |
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play file.xxx flanger 0.6 0.87 3.0 0.9 0.5 -t |
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If the decay is a bit lower, than the effect sounds more |
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popular: |
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play file.xxx flanger 0.8 0.88 3.0 0.4 0.5 -t |
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The drunken loudspeaker system: |
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437 |
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438 |
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439 |
play file.xxx flanger 0.9 0.9 4.0 0.23 1.3 -s |
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440 |
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441 |
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442 |
__Reverb__ |
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443 |
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444 |
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445 |
The reverb effect is often used in audience hall which are |
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446 |
to small or contain too many many visitors which disturb |
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447 |
(dampen) the reflection of sound at the walls. Reverb will |
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448 |
make the sound be perceived as if it were in a large hall. |
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449 |
You can try the reverb effect in your bathroom or garage or |
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450 |
sport halls by shouting loud some words. You'll hear the |
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451 |
words reflected from the walls. |
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452 |
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453 |
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454 |
The biggest problem in using the reverb effect is the |
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455 |
correct setting of the (wall) delays such that the sound is |
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456 |
realistic and doesn't sound like music playing in a tin can |
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457 |
or has overloaded feedback which destroys any illusion of |
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458 |
playing in a big hall. To help you obtain realistic reverb |
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459 |
effects, you should decide first how long the reverb should |
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460 |
take place until it is not loud enough to be registered by |
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461 |
your ears. This is be done by varying the reverb time |
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462 |
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463 |
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464 |
Since audience halls do have a lot of walls, we will start |
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465 |
designing one beginning with one wall: |
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466 |
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467 |
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468 |
play file.xxx reverb 1.0 600.0 180.0 |
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469 |
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470 |
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471 |
One wall more: |
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472 |
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473 |
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474 |
play file.xxx reverb 1.0 600.0 180.0 200.0 |
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475 |
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476 |
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477 |
Next two walls: |
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478 |
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479 |
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480 |
play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 |
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481 |
240.0 |
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482 |
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483 |
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484 |
Now, why not a futuristic hall with six walls: |
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485 |
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486 |
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487 |
play file.xxx reverb 1.0 600.0 180.0 200.0 220.0 240.0 280.0 |
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300.0 |
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489 |
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490 |
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491 |
If you run out of machine power or memory, then stop as many |
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492 |
applications as possible (every interrupt will consume a lot |
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493 |
of CPU time which for bigger halls is absolutely |
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494 |
necessary). |
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495 |
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496 |
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497 |
__Phaser__ |
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498 |
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499 |
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500 |
The phaser effect is like the flanger effect, but it uses a |
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501 |
reverb instead of an echo and does phase shifting. You'll |
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502 |
hear the difference in the examples comparing both effects |
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503 |
(simply change the effect name). The delay modulation can be |
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504 |
sinusoidal or triangular, preferable is the later for |
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505 |
multiple instruments. For single instrument sounds, the |
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506 |
sinusoidal phaser effect will give a sharper phasing effect. |
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507 |
The decay shouldn't be to close to 1.0 which will cause |
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508 |
dramatic feedback. A good range is about 0.5 to 0.1 for the |
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509 |
decay. |
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510 |
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511 |
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512 |
We will take a parameter setting as for the flanger before |
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513 |
(gain-out is lower since feedback can raise the output |
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514 |
dramatically): |
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515 |
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516 |
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517 |
play file.xxx phaser 0.8 0.74 3.0 0.4 0.5 -t |
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518 |
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519 |
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520 |
The drunken loudspeaker system (now less |
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521 |
alcohol): |
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522 |
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523 |
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524 |
play file.xxx phaser 0.9 0.85 4.0 0.23 1.3 -s |
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525 |
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526 |
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527 |
A popular sound of the sample is as follows: |
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528 |
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529 |
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530 |
play file.xxx phaser 0.89 0.85 1.0 0.24 2.0 -t |
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531 |
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532 |
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533 |
The sample sounds if ten springs are in your |
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534 |
ears: |
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535 |
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536 |
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537 |
play file.xxx phaser 0.6 0.66 3.0 0.6 2.0 -t |
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538 |
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539 |
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540 |
__Compander__ |
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541 |
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542 |
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543 |
The compander effect allows the dynamic range of a signal to |
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544 |
be compressed or expanded. For most situations, the attack |
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|
545 |
time (response to the music getting louder) should be |
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546 |
shorter than the decay time because our ears are more |
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|
547 |
sensitive to suddenly loud music than to suddenly soft |
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|
548 |
music. |
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549 |
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550 |
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551 |
For example, suppose you are listening to Strauss' |
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552 |
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553 |
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554 |
play file.xxx compand 0.3,1 -90,-90,-70,-70,-60,-20,0,0 -5 0 |
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555 |
0.2 |
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556 |
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557 |
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558 |
The transfer function ( |
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|
559 |
very'' soft sounds between -90 and -70 decibels (-90 is |
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|
560 |
about the limit of 16-bit encoding) will remain unchanged. |
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|
561 |
That keeps the compander from boosting the volume on |
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|
562 |
'' |
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563 |
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564 |
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565 |
__Changing the Rate of Playback__ |
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|
566 |
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|
567 |
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|
568 |
You can use stretch to change the rate of playback of an |
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|
569 |
audio sample while preserving the pitch. For example to play |
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|
570 |
at 1/2 the speed: |
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571 |
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572 |
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573 |
play file.wav stretch 2 |
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574 |
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575 |
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576 |
To play a file at twice the speed: |
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|
577 |
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578 |
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|
579 |
play file.wav stretch .5 |
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|
580 |
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|
581 |
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|
582 |
Other related options are |
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|
583 |
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|
584 |
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585 |
play file.wav speed 2 |
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586 |
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587 |
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|
588 |
To raise the pitch of a sample 1 while note (100 |
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|
589 |
cents): |
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|
590 |
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591 |
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|
592 |
play file.wav pitch 100 |
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|
593 |
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594 |
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595 |
__Other effects (copy, rate, avg, stat, vibro, lowp, highp, |
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|
596 |
band, reverb)__ |
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|
597 |
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|
598 |
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|
599 |
The other effects are simple to use. However, an |
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|
600 |
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|
601 |
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|
602 |
__More effects (to do !)__ |
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|
603 |
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|
604 |
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|
605 |
There are a lot of effects around like noise gates, |
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|
606 |
compressors, waw-waw, stereo effects and so on. They should |
|
|
607 |
be implemented, making SoX more useful in sound mixing |
|
|
608 |
techniques coming together with a great variety of different |
|
|
609 |
sound effects. |
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|
610 |
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|
|
611 |
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|
612 |
Combining effects by using them in parallel or serially on |
|
|
613 |
different channels needs some easy mechanism which is stable |
|
|
614 |
for use in real-time. |
|
|
615 |
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|
|
616 |
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|
|
617 |
Really missing are the the changing of the parameters and |
|
|
618 |
starting/stopping of effects while playing samples in |
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|
619 |
real-time! |
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|
620 |
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|
621 |
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|
622 |
Good luck and have fun with all the effects! |
|
|
623 |
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|
624 |
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|
625 |
Juergen Mueller (jmueller@uia.ua.ac.be) |
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|
626 |
!!SEE ALSO |
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|
627 |
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|
628 |
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|
629 |
sox(1), play(1), rec(1) |
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|
630 |
!!AUTHOR |
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|
631 |
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|
632 |
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|
633 |
Juergen Mueller (jmueller@uia.ua.ac.be) |
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|
634 |
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635 |
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|
636 |
Updates by Anonymous. |
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|
637 |
---- |