lolcommits

Every now and again I look at the lolcommits gem, this allows a git user to take a snapshot each time that they commit code to their repository.

The only bit it took me a while to work out was the correct way to get the twitter option to post.

After some playing around I discovered the following sequence usually works:

lolcommits -p twitter --config

and type true when you get the prompt

enabled:

Once you have committed to your repo, then type

lolcommits -c

this will take a fresh screenshot from your latest commit and give you a url to authorise the tweet activity. Enter the url into a browser, then paste the code into the command prompt and all your future commits will be sent to annoy your followers.

Examples of this hideous idea are here (how many of these places do you recognise?):

Delays, Combs and Pitch

Introduction

Today I am going to talk a about the delay spectrum, showing examples of comb filtering, creating pitches with delays, slapback delays and synchronized long delays.

The Delay Spectrum

Terms used in delay effects

The delay time is how long the signal is delayed before being mixed in with the original signal.

We have dry and wet signals in filtering. A wet signal is the delayed signal. The dry signal is the undelayed signal.

Feedback is how much of the wet signal is returned back to itself. This is a gain stage and increases the amount of feedback that can be heard, and the time that the signal will continue to be heard.

Comb Filtering

Comb Filter Frequency Response Diagram

A comb filter adds a delayed version of a signal to itself, causing constructive and destructive interference. The frequency response of a comb filter has a series of evenly spaced spikes, making it look similar to a comb.

In this diagram, you can see the comb like signal. There is a 1mS delay between the original signal and both wet and dry signals are at 0dB. Where the two signals peak together, you can see that signal have been increased by 6dB. Where the signals are opposite there is around a 100dB notch.

Creating Pitches with Delays

With feedback applied and a 1mS delay, a clicking signal will appear to have a pitch caused by the addition of the dry signal with the fed back wet signal. If the delay is slowly increased, the pitch will appear to have a lower frequency. This relationship between the delay and the percieved pitch is a core part of generating electronic music.

Slapback Delay

A slapback delay has a medium delay time, typically between 75 and 250mS with very little or no feedback. This gives the effect of having sound reflecting from a wall with no additional echo's, or in other words making a similar sound to a room.

Synchronised Long Delays

A long delay gives a distinct echo to the sound, that matches what we would find in nature. It is important to synchronise the delay to the underlying musical content, in time with the quater notes for example.

To aid the listener, the wet signal needs to sound distinct to the dry original signal. A long delay can clash with harmonies and complex rhythms, so long delays need to be applied carefully and only after ensuring that the delay enhances the musical experience.

Reflection

Although a delay is a very simple concept, in actuality they are very complex effects to set up and make a usable and beneficial improvement to a song.

I would like to thank those of you that have read through and commented on my work, and wish you well with your coursework. I am finding it hard to believe that we're rapidly approaching the end of this course!

Dynamic Range

Introduction

Welcome to my week 4 assignment for the Music Production course. Today I am going to look at Dynamic Range and some methods a producer can use to manipulate it.

Dynamic Range

This is simply the difference between the loudest and the quietest sounds in a recording.

The musicians give a natural dynamic range by following their score and playing louder or softer during their performance. The post-production work adjusts the dynamic range to give more feel to the music.

There are a number of areas where dynamic range can be adjusted:

• Macro Scale>
• Transients
• Compression
• Expansion
• Dynamic Processors

The Macro Scale

The producer will look at the overall performance, and will see the relative levels between sections of the recording. They will ride the fader and automate the volume levels to strengthen the difference between verses and the chorus.

Transients

A transient is where the amplitude changes a lot during a short timescale. Examples of transient sounds include snare hits.

Compression

Compression reduces the dynamic range and brings volume levels to all the parts of the music into a similar level.

Expansion

Expansion increases the dynamic range, and gives more emphasis to sections of the music, either by bringing the perceived volume level up or down.

Dynamic Processors

A producer will use a dynamic processor to adjust the level of the audio following rules. Expanders, gates, compressors and limiters are types of dynamic processors.

Threshold

The point that a dynamic processor starts taking effect is known as the threshold. This is measured in dB, and a lower threshold (eg -30dB) means a large portion of the signal will be affected compared to a higher threshold of -5dB.

Ratio

The ratio is how much the processor changes the level once the threshold has been passed. The higher the ratio, the more the signal will change.

A ration of 6:1 means that if the input level is 6dB over the threshold, the output signal will be 1dB over threshold. The gain in this case has been reduced by 5dB.

Attack and Release

Attack and release provide a measure of control over how quickly a compressor acts.

The attack phase is the time when the compressor is decreasing gain to match the level that is set by the ratio. The release phase is when the compressor is increasing gain once the level has fallen below the threshold.

A compressor's attack and release are measured in mS, the time that it takes the gain to change a set amound of dB.

Soft and Hard Knees

The knee is how quickly the effect takes place. A soft knee slowly increases the compression ratio as the level increases and eventually reaches the compression ratio set by the user. The softer knee's reduces the audible change from uncompressed to compressed, especially for higher ratios.

Compression

A downward compressor reduces loud sounds over a certain threshold while quiet sounds remain unaffected.

An upward compressor will increase the loudness of sounds below the threshold while leaving louder sections unchanged.

Both downward and upward compression reduce the dynamic range of an audio signal.

Expanders

An expander increases the dynamic range of an audio signal, and typically are used to make quiet sounds even quieter by reducing the level of the signals that fall below a set threshold level.

A noise gate is a type of expander.

Reflection

I have given a brief introduction to Dynamic Range and the ways that a producer can change the level of a song.
The one thing the producer must consider about above all else is that the essential character of the music is not changed.

I would like to thank you for taking the time to read my discussion on Dynamic Range. There is so much to learn about the topic, and it will take a lifetime to fully understand the practical aspects.

Mixing Desks - The Channel Strip

Introduction

Alto ZMX164FX USB Mixing Desk

Today I am going to show you how signals move through a mixer, using the Alto ZMX164FX USB analogue mixing board pictured to the right to demonstrate.

Mixing boards look really complicated, but they are modular devices and contain a number of repeated sections.

This board has 12 channel strips, four subgroups, a special effects section, the monitor section, and the main mix fader.

The sections of the board are shown in the diagram to the right.

This lesson is going to focus on the signal flow through the channel strip. I am also going to briefly discuss where the signals go after leaving the channel strips - the sub and main mixes.

The Channel Strip

As a rule, signals flow from the top to the bottom of the channel strip, although there are a couple of exception to this rule, they will be described later.

I am going to describe the mono input channels 1-8, on this mixer channels 9-16 are stereo inputs and are have slightly different operation.

Although the inputs are mono, the output after the fader are a stereo signal.

The Input

Inputs

The signal starts with the microphone or line input at the top. The line inputs are TRS balanced/unbalanced inputs.

The insert is to allow connecting an external effects rack into the mix using a TRS connector.

The line gain allows the level to be adjusted to line level. This is different to a DAW, as the DAW expects the signals to already be at line level.

The peak led illuminates if the signal should rarely, if ever, light otherwise you will have distorted signals from the very start.

The low-cut button switches in a 75Hz low frequency filter to reduce hum from a mains power supply or stage rumbles.

The Equaliser

Signal Routing

The blue knobs are a 3-band EQ with a sweepable MID range.

Aux Sends

The Aux Sends can be thought of as a separate mix

The yellow knobs are the auxiliary sends level controls. These control the signal levels to the two auxiliary bus.

AUX1 and AUX2 can be switched to pre or post fader and can be used for monitor or inputs to an effects and sound processor.

AUX 3 and AUX4 are configured as post faders. AUX4 can be assigned to the onboard effects module.

The PAN control is used to position the signal to the left or right side. In the central position the audio will appear in the centre of the stage. This is also an out of order signal flow.

The Fader Section

Channel Fader
and Options

At the top of the fader section is a peak warning led. Ignore this at your peril, it flashes to warn you when you are reaching signal saturation and possible distortion.

The mute button is equivalent to pulling the fader down, and causes the mute led to illuminate.

The channel can route to sub 1-2, sub 3-4, and the Main LR busses.

The solo button routes the signal from the channel to the control room.

Finally, the fader controls the overall level of the signal sent out to the main and sub busses. On this board, the unity gain is marked with a 0. The fader allows from -∞ to +15 db. The ∞ means that effectively no signal will be heard from the channel.

Ideally you want to keep signals around unity as much as possible, this gives the least risk of distortion occurring in your work.

The DSP

DSP Options

The mixer has a 24 bit digital special effects unit that can add flangers, reverb, chorus, delay and combinations of these. There are 16 preset options for each effect selected.

The effects out control is the signal sent to DFX OUT and can be set between -∞ to +10 db.

The AUX1 and AUX2 controls are used to set the signal level from AUX Return 4 on the channel strip to the AUX Send1 and 2. The range is between -∞ to +15 db.

The Peak LED illuminates if the input signal is too strong. The DSP can be muted, and this causes the peak LED to illuminate.

The board power and phantom power indicators can also be seen on this picture.

The Equaliser

Graphic Equaliser

The board has a nine-band graphic equaliser, and a bypass button.

This allows the fine-tuning of the overall output audio levels. 

Sub groups and Main Mix

Subgroups and Main Mix Faders

This is where the output from all of the channels appears.

The four white sub faders control the levels for each of the sub channels.

The red fader is the main mix, the output that your audience are listening to. This board controls the overall level, unlike some that can control the individual left and right levels.

The channel that each of the sub groups appears on the main mix is controlled by the assign to main mix buttons.

Subgroup assignment

Pressing the LEFT button will assign the subgroup to the left main mix, the RIGHT sends to the right  main mix, and having both selected will send to both channels.

Reflection

I have focussed on the parts of the board that I have used, the only section I haven't discussed are the auxiliary sends and returns. Apart from the built in DSP, I don't have any equipment to connect and experiment with (donations gratefully accepted!)

Creating the lesson has helped me understand the operation of my board in more detail, in particular how the special effects unit operates.

I would like to thank you for your time in reading my lesson on the Channel Strip, and I hope you have gained as much as I have (so much so, that I'm tempted to create lessons on some of the other options for this assignment).

Now I want to go and put theory into practice - apologies to my neighbours!

Preparing a Project in a DAW

Introduction

This is my second assignment for the Introduction to Music Production course. I am going to look at setting up a project in a Digital Audio Workstation for recording.

I am going to use Audacity and Reaper for this demonstration.

Audacity is a free and open source application that is available for Windows, Linux and the OSX environments. I am using the OSX version of Audacity, the other versions look almost the same.

Reaper is a commercial product for OSX, I am currently using the evaluation version of the software.

I was hoping that version 3.0 of Ardour would have been released for OSX in time to do a comparison over three packages, however currently only the Linux version was available.

Setting up a DAW Project

Preproduction Checklist

Before we start our project, we need to consider the following things:

• Where and What
• Digital Audio Preferences
• File Type
• Hardware Settings
• Buffer Size

Saving your files in suitable locations  and choosing appropriate filenames makes it much easier to locate, edit and backup your work. You do back it up, don't you?

Audacity

Audacity has a friendly feel to the interface and is very easy and simple to use.

Saving Work

Folders

You can see I am keeping materials for this course organised into folders by assignment.

Audio Preferences

Sample Rate and Size

I have set the sample rate to 44.1kHz and the sample size to 24-bit.

This is a higher rate and larger word size than used in CD quality recordings.

File Type

Audacity uses it's own default file type that keeps all data about the recording in a project file and folder.  It is possible to name individual tracks on the recording.

Hardware Settings

Recording hardware selected

I am recording from my mixing desk, it shows as USB Audio CODEC in the lists.

Buffer Size

Buffer Size

Audacity gives a buffer size in mS, not the number of samples and I am leaving at the default 100mS.

Reaper

Reaper has far more options than Audacity, and provides a more professional seeming interface, although not all of the options match up with the descriptions given in the lectures.

Saving Work

I have selected a 'ReaperRecordings' folder for saving the files produced by Reaper.

Audio Preferences

I have selected a 44.1 kHz sample rate with a 24 bit sample size for the recordings.

 

File Type

Project Save Settings

I have set Reaper to use a WAF format for saving, there are several other options available.

Hardware Settings

Hardware Settings

I am using the USB CODEC for my mixing desk.

Reaper has an option to request a sample rate, the default was 48,000.

Buffer Size

Buffering Options

Reaper does not seem to have an option for the buffer size in the same way as the lectures suggest. 

Reflections

As with everything involved with computers, there are many different ways of describing and doing the same tasks. Using a DAW is much the same with the two packages looked at having a different look and feel.

I'd like to thank the reviewers for taking the time to read through my resources, I hope you gain as much from the tasks as I have while preparing this.

Songwriting Week 2

This week has been looking at number of lines to create a feel within a song.

The assignment was on having a stable chorus with an unstable verse. I'm not going to post my efforts here until after the assignment has been graded, and depending on the reviews I might not even then! The title is "The Runaway" and is further explorations on the theme.

I started looking at the idea of the song a few months ago (I was at the Ketton Ox in Yarm near Middlesbrough at an open acoustic night when the idea struck me), and have had two or three attempts at putting words together for it

I'm still finding the unit interesting, but really need to organise my time better; especially as I have another course starting in April.

Coming up with ideas for lyrics may be something that I'm capable of, but learning enough musical theory and practice to create the accompanying melody is going to be a little more challenging. I also need to make more time to continue learning the guitar, I feel that I've barely picked it up in the last month or so.

Now to look at my assignment for the Music Production course that is also due soon...

Music Production week 2

I've decided that I'm not a fan of working Saturdays with a day off mid-week.

It probably wasn't helped that this last few days have been particularly busy for me, with going to an event in Newcastle after the Cub Scout meeting on Friday, followed by working Saturday then going to a party on Saturday night.

I have caught up with the Music Production videos early this week, and have completed the quizzes on a sensible day for a change. I just have to do the reviews for the week 1 assignment and prepare my assignment for week 2.

This weeks lectures have been about making edits using a DAW and how the midi controllers send messages. Although again there were lots of videos to watch, they were only a minute or so each and worked very well in the small bites. It gives time to have a try at the ideas and techniques using the software that you have on your own machine.

I am planning on leaving the assignment until Wednesday before starting, I seem to have a free evening and intend to make good use of it to so more preparation for the second assignment. Although I might also make good use of it by seeing Lynn for a couple of hours.

• Prepare a project in your DAW using the project checklist from the material as your guidelines.
• Record audio in your DAW including preparing the project, creating the track(s), setting the click and countoff, and recording efficiently.
• Perform the important editing tasks in your DAW including: trim, separate, crossfade, merge, grid, cycle, markers, zoom, name and color.
• Add a software instrument and record MIDI and quantize in your DAW. Including preparing the track(s), adding the instrument, setting the click and countoff, and recording efficiently.
• Efficiently create a compile from multiple audio recordings in your DAW.
• The Analog to Digital conversion process.
• Editing an imperfect audio performance to correct the timing.

I am looking at three DAW's at the moment, and will need to choose one to focus on. More decisions!