Jey's Podcasting Journal

I'm taking a pause to reflect and celebrate this fiftieth episode. 

My top three lessons are:

1. Deadlines are essential to keeping my promise to ship. 

2. Creating a clear mission I believe in is also key to help me keep my promises to you, my listener. 

3. Falling in love with the process, is a greater reward than how an episode turns out

Recording Mic: Zoom LMF-2 Lavalier mic

Recording device : Zoom F2 32 bit float Field Recorder

Recording location: My Car

Recording format: WAV (32 bit float mono)

Editing app (and EQ) : Audacity

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Episode art: Canva

Anchor automatically converts the WAV file into a 128kbps, stereo .m4a file

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I have to turn off my workflow to present this episode on generating noise to demonstrate the concept of Noise floor. So this episode is going to sound very rough. 

Learn how to generate noise so you  can calibrate a noise floor level that's acceptable to you. 

Also an intro to the Youlean Loudness meter (free version). A great way to measure loudness. 

And you can use LUFS to determine your noise-floor too. So pick a method that works for you and stick to it. 

I also use the Youlean meter to make sure my published episodes are at the correct level for mono files (-19LUFS) 

The Youlean Loudness Meter can be found here: https://youlean.co/youlean-loudness-meter/

PRODUCTION NOTES

Recording Mic: Zoom LMF-2 Lavalier mic

Recording device : Zoom H6 hooked up to a Dell Laptop in mix-minus configuration. (I got some serious RF interference, from the laptop, which I removed. I kept in all the other noise). 

Recording location: Dining table

Recording format: WAV (24 bit mono)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: None (ouch!!!)

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Episode art: Canva

Anchor automatically converts the WAV file into a 128kbps, stereo, .m4a file

The noise in your recording environment in addition to the noise in your recording equipment adds up to what is called the noise floor. Let's explore how to measure this. Understand what an acceptable noise floor level might be. 

And understand what a signal chain is and how to use it to systematically reduce the noise floor. 

And there's just enough time to explore what room tone is and why it should be captured in all your recordings. 

Note: my experiment at the end of the recording didn't work. It didn't sound as jarring because Auphonic.com cleaned up the noise so well in the rest of the tape that the gaps had no impact!!! Oh well. Try the experiment for yourself (before running your tape through Auphonic). 

PRODUCTION NOTES

Recording Mic: Zoom LMF-2 Lavalier mic

Recording device : Zoom F2 32 bit float Field Recorder

Recording location: Standing into my closet

Recording format: WAV (32 bit float mono)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Episode art: Canva

Anchor automatically converts the WAV file into a 128kbps .m4a file

With 32bit Float recorders you'll never have to set the gain again. Never have to worry about clipping. Or having guests that speak to quietly. You don't even have to monitor your recorder. Because you can fix everything in post-production. Sound to good to be true? Join me and find out!

To dig into the mathematics behind 32bit float technology check out this site: https://www.sounddevices.com/32-bit-float-files-explained/

PRODUCTION NOTES

Recording Mic: Zoom LMF-2 Lavalier mic

Recording device : Zoom F2 32 bit float Field Recorder

Recording location: Standing into my closet

Recording format: WAV (32 bit float mono)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a 128kbps .m4a file

This process makes sure our tape is at the right volume. But what does right mean. What's the difference between Peak Normalization and Loudness Normalization. 

Which is more important for podcasting? 

Let's Explore. 

And thanks to the Audacity to Podcast show. I drew a lot of info from this site and this post in particular: https://theaudacitytopodcast.com/why-and-how-your-podcast-needs-loudness-normalization-tap307/

Tools for loudness normalization:

Auphonic.com

FixMyLevels 

PRODUCTION NOTES

Recording Mic: Samson Q2U with foam wind cover and pop-filter for the test. 

Recording device : Zoom H6

Recording location: Sitting on my bed.

Recording app: Audacity.

Recording format: WAV (16 bit mono)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a stereo 128kbps .m4a file

Too much sibilance makes listening uncomfortable. Let's explore what we can do to reduce this during production including the "pencil trick" for side addressed mics. 

And in post production, let's explore why it's more efficient to use a plugin than try to manually remove this type of distortion. 

PRODUCTION NOTES

Recording Mic: Samson Q2U no wind cover or pop-filter for the test. Both put back on for the narration. 

Recording device : Zoom H6 

Recording location: Sitting on my bed. 

Recording app: Audacity.

Recording format: WAV (16 bit mono)

Sample rate: 44100Hz

Editing app: Audacity 

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a stereo 128kbps .m4a file

The order that you apply your effects and treatments matter. Join me to explore this. This is a very quick episode. I want you to understand the logic of the sequence I'm presenting this material so you can experiment for yourself. 

Remember: Remove before Enhancing. 

PRODUCTION NOTES

Recording Mic: Microsoft Lifechat LX6000 USB Headset

Recording device : Dell G7 Windows 10 via USB

Recording location: Sitting at dining table - no pillowfort 

Recording app: Audacity for Windows

Recording format: WAV (16 bit stereo)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: EQ applied in Audacity. Then cleanup and Loudness (-19LUFs) applied at Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a 128kbps .m4a file

Today I test the in-built noise suppression tool in Zoom. And I test a commercial noise suppression tool called Krisp. Let's see if we can save time in post-production by using simple tools to suppress noise during recording. 

PRODUCTION NOTES

Recording Mic: Samson Q2U for the narration. Microsoft Lifechat LX 6000 for noise suppression tests. 

Recording device : Zoom H6 for the narration. Dell G7 Windows 10 aptop for the tests. 

Recording location: Narration recorded sitting on my bed. Test done in my living room 10 feet (3m) apart.  

Recording app: For the narration I used Audacity.

Recording format: WAV (16 bit stereo)

Sample rate: 44100Hz

Editing app: Audacity 

Sound Treatment: None. I used Audacity to normalize to -19LUFS

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a 128kbps .m4a file

How do you get rid of sounds that leak onto your mic when you are not speaking? A noise gate is the answer. 

It's the go to tool if you or or guest's mics are bleeding into each other's recordings. 

It's good for removing loud breathing too. Let's explore how to use this tool. 

PRODUCTION NOTES

Recording Mic: MPOW 071 USB & 3.5mm Headset

Recording device : Samsung Galaxy Note 4 for the narration. 2 x Samson Q2U into a Zoom H6 for the bleed tests. 

Recording location: Narration recorded sitting on my bed. Bleed test done in my son's bedroom we were 8 feet (2.5 meters) apart.  

Recording app: For the phone I used Auphonic Edit. 

Recording format: WAV (16 bit stereo)

Sample rate: 44100Hz

Editing app: Audacity (including for the Noise Gate) 

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a 128kbps .m4a file

If your sound is amplified too high then you'll go beyond your recorder's ability to capture all the frequencies resulting in a distortion called clipping. Let's explore what we can do about that. 

PRODUCTION NOTES

Recording Mic: Samson Q2U Dynamic with pop filter and foam windshield.

Recording device : Zoom H6

Recording location: Sitting at my son's desk in the corner of his bedroom. Two pillows in a v shape around the mic. 

Recording format: WAV (16 bit stereo)

Sample rate: 44100Hz

Editing app: Audacity

Sound Treatment: Auphonic Productions at Auphonic.com

Auphonic.com settings [to output as a 16bit mono WAV file]: Adaptive Leveler (on), Loudness Normalization (Loudness Target -19 LUFS ), Filtering (on), Noise and Hum Reduction (Reduction Amount: Auto), Sample rate: 44.1kHz

I uploaded the resulting WAV file to Anchor.fm to host the file and added the music from their free music stock.

Anchor automatically converts the WAV file into a 128kbps .m4a file