woudsma · April 11, 2023 21:27 · Apr 11, 2023 · Apr 11, 2023 · Apr 11, 2023 · Apr 11, 2023
diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -9,11 +9,11 @@ This guide is for UNIX (MacOS or Linux) users, the process would be similar on W
 ffmpeg -i $INPUT -acodec pcm_s16le -ac 1 -ar 48000 $OUTPUT
 
 # -i: This parameter specifies the input file to be processed.
-# $INPUT: This is a placeholder for the actual input file name.
+# $INPUT: This is a placeholder for the actual input file path.
 # -acodec pcm_s16le: This parameter specifies the audio codec to be used for the output file. In this case, pcm_s16le is a codec that specifies uncompressed 16-bit audio data.
 # -ac 1: This parameter specifies the number of audio channels to be used in the output file. In this case, 1 specifies a mono audio output.
 # -ar 48000: This parameter specifies the audio sampling rate to be used in the output file. In this case, 48000 specifies a sampling rate of 48 kHz.
-# $OUTPUT: This is a placeholder for the actual output file name.
+# $OUTPUT: This is a placeholder for the actual output file path.
 ```
 
 ---

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -9,7 +9,7 @@ This guide is for UNIX (MacOS or Linux) users, the process would be similar on W
 ffmpeg -i $INPUT -acodec pcm_s16le -ac 1 -ar 48000 $OUTPUT
 
 # -i: This parameter specifies the input file to be processed.
-# $INPUT: This is a placeholder for the actual input file name. It is likely that this command is part of a script or batch file, and the $INPUT variable is being replaced with the actual input file name at runtime.
+# $INPUT: This is a placeholder for the actual input file name.
 # -acodec pcm_s16le: This parameter specifies the audio codec to be used for the output file. In this case, pcm_s16le is a codec that specifies uncompressed 16-bit audio data.
 # -ac 1: This parameter specifies the number of audio channels to be used in the output file. In this case, 1 specifies a mono audio output.
 # -ar 48000: This parameter specifies the audio sampling rate to be used in the output file. In this case, 48000 specifies a sampling rate of 48 kHz.

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -4,6 +4,18 @@ Samples need to be 16 bit / 48kHz mono wav files.
 - [Zsh](https://github.com/ohmyzsh/ohmyzsh/wiki/Installing-ZSH) (run `zsh --version` to check if it's installed already)
 - [FFmpeg](https://ffmpeg.org/) (`brew install ffmpeg`)
 
+This guide is for UNIX (MacOS or Linux) users, the process would be similar on Windows machines with FFmpeg though. The required FFmpeg options are:
+```sh
+ffmpeg -i $INPUT -acodec pcm_s16le -ac 1 -ar 48000 $OUTPUT
+
+# -i: This parameter specifies the input file to be processed.
+# $INPUT: This is a placeholder for the actual input file name. It is likely that this command is part of a script or batch file, and the $INPUT variable is being replaced with the actual input file name at runtime.
+# -acodec pcm_s16le: This parameter specifies the audio codec to be used for the output file. In this case, pcm_s16le is a codec that specifies uncompressed 16-bit audio data.
+# -ac 1: This parameter specifies the number of audio channels to be used in the output file. In this case, 1 specifies a mono audio output.
+# -ar 48000: This parameter specifies the audio sampling rate to be used in the output file. In this case, 48000 specifies a sampling rate of 48 kHz.
+# $OUTPUT: This is a placeholder for the actual output file name.
+```
+
 ---
 
 Helper function to convert an audio sample to Alpha Base compatible format:

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -48,7 +48,7 @@ for f in ./**/*.wav; do mv "$f" ${f//-16bit-48khz-mono/}; done
 
 Or optionally do it all at once. I personally like the above approach because I want to check things before removing/renaming/etc.
 
-I strongly recommend to make a copy/backup of the folder with the wavs that you want to convert. This script removes the old wavs with the `rm $f` command.
+I strongly recommend to make a copy/backup of the folder with the wavs that you want to convert. This script removes the original wavs with the `rm $f` command after converting them.
 ```sh
 # Go to your samples directory
 cd /path/to/samples-directory

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -47,6 +47,8 @@ for f in ./**/*.wav; do mv "$f" ${f//-16bit-48khz-mono/}; done
 ---
 
 Or optionally do it all at once. I personally like the above approach because I want to check things before removing/renaming/etc.
+
+I strongly recommend to make a copy/backup of the folder with the wavs that you want to convert. This script removes the old wavs with the `rm $f` command.
 ```sh
 # Go to your samples directory
 cd /path/to/samples-directory

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -34,7 +34,7 @@ rm ./**/*.asd
 
 # Convert all wavs to 16 bit / 48kHz mono wavs
 for f in ./**/*.wav; do wav2mono16bit48khzwav $f; done
-# Or:
+# Or if you have both mp3 and wav files:
 for f in ./**/*.{mp3,wav}; do wav2mono16bit48khzwav $f; done
 
 # Remove the old (non-mono) wavs

diff --git a/jomox-alphabase-convert-samples-ffmpeg.md b/jomox-alphabase-convert-samples-ffmpeg.md
@@ -0,0 +1,63 @@
+# Converting samples before import to JoMoX Alpha Base
+Samples need to be 16 bit / 48kHz mono wav files.  
+#### Requirements:
+- [Zsh](https://github.com/ohmyzsh/ohmyzsh/wiki/Installing-ZSH) (run `zsh --version` to check if it's installed already)
+- [FFmpeg](https://ffmpeg.org/) (`brew install ffmpeg`)
+
+---
+
+Helper function to convert an audio sample to Alpha Base compatible format:
+
+> Tested only with `zsh`. Add this function to your `~/.zshrc` file for example (and reload with `source ~/.zshrc`)
+
+```sh
+function wav2mono16bit48khzwav() {
+  INPUT=$1
+  OUTPUT=${INPUT:r}-16bit-48khz-mono.wav
+  ffmpeg -i $INPUT -acodec pcm_s16le -ac 1 -ar 48000 $OUTPUT
+}
+```
+Usage: `wav2mono16bit48khzwav /path/to/sample.wav`  
+Creates a file next to the original file named `sample-16bit-48khz-mono.wav`  
+_Should_ also work with mp3 and other audio files as input
+
+---
+
+To convert an entire directory including subdirectories with samples:
+> Note that the Alpha Base doesn't support subdirectories, all samples must either be placed in the SD card root, or you can store samples in directories. Those can't have subdirectories, so samples should be placed max 1 level deep
+```sh
+# Go to your samples directory
+cd /path/to/samples-directory
+
+# (optional) Remove any .asd Ableton files if you want
+rm ./**/*.asd
+
+# Convert all wavs to 16 bit / 48kHz mono wavs
+for f in ./**/*.wav; do wav2mono16bit48khzwav $f; done
+# Or:
+for f in ./**/*.{mp3,wav}; do wav2mono16bit48khzwav $f; done
+
+# Remove the old (non-mono) wavs
+for f in ./**/*.wav; do test "${f#*16bit-48khz-mono}" = "$f" && rm $f; done
+
+# Rename all samples by removing "-16bit-48khz-mono" from the filename
+for f in ./**/*.wav; do mv "$f" ${f//-16bit-48khz-mono/}; done
+```
+
+---
+
+Or optionally do it all at once. I personally like the above approach because I want to check things before removing/renaming/etc.
+```sh
+# Go to your samples directory
+cd /path/to/samples-directory
+
+# Convert and replace all samples with 16 bit / 48kHz mono wavs
+for f in ./**/*.wav; do
+  OUTPUT=${f:r}-16bit-48khz-mono.wav
+  ffmpeg -i $f -acodec pcm_s16le -ac 1 -ar 48000 $OUTPUT && rm $f && mv $OUTPUT ${f:r}.wav
+done
+```
+
+---
+
+Now you can copy the samples to the SD card for use on the Alpha Base 👍
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