Podcast sample rate
 

Hi all

I made a recording of a conference yesterday using 16Bit 48kHz sampling. I have been asked to create podcasts from the recording and I am seeing from the internet that there has at some time been a preference for 44.1 kHz sammpling due to incompatibilities with adobe/macromedia flash.

My question is: is it still the case that 44.1kHz is the recommended sampling rate for podcasts? If so, what is the least worst method for converting my files (I have Sony Vegas).

Cheers, Jon

I'm seeing that 22.05kHz is sufficient for speech, and has the added benefit of smaller file sizes. Any thoughts on artefacts when goinbg form 48kHz to 22.05kHz? Jon

Most of the podcasts I subscribe to deliver as mono 44.1k 64kbps mp3s, so I'd suggest starting with that setting. With the terrible little earbuds or $5 computer speakers most people listen to podcasts with, I wouldn't worry too much about artifacts in the conversion.

How many clips do you have? If it's just a few, then I'd simply render them from your Vegas timeline to the output file of your choice. Vegas does a good job quality-wise in rendering to a setting that differs from the editing timeline but if you have lots of clips already edited and saved, a batch processor would obviously be easier. I don't have any recommendations for a free or low-cost batch processor but I'm sure there are people here who've got a favorite.
For my last podcast, which started out as an audio-only project, I recorded at 16-bit 44.1k wav. Later it morphed into a video podcast so I added in the PowerPoint slides (as bmp files) and other graphics as well as intro and exit music. Rendered to mp4 for the video version and 44.1k 128kbps mp3 for the audio-only version. It sounded very clean to me but the 15:39 audio-only file was right at 15Mb in size.

This site could be helpful ...

Tutorials.

Cheers.

Hi Guys

Thanks for the comments, I'm just back from a weekend away so I shall check out the website soonest. The files are not yet edited so I will do that in Vegas first before testing the conversion.

Cheers, Jon

If you do not need a stereo MP3, a mono file will be of equal data compression quality at half the bitrate .
For instance: A 128kbps stereo file would have roughly the same quality as 64kbps mono file, all parameters being equal except mono/stereo. Keep in mind, this does not work the same way as PCM files, rendering a mono version half's the file size.
I have gotten 'decent' quality with 'acceptable' artifacts using a constant sample rate (CBR) of 20kbps @ 11.025khz. for dial-up connection users. If you are streaming, it is recommended using CBR .

44.1k at 64kbps in mono is pretty much standard. If you're tempted to go stereo, I do recommend you look into VBR encoding. At 50%, it's roughly equivalent to 64kbps mono and only increases bitrate when there is stereo separation such as music.

Thanks all. I just finished the project in mono at 44.1kHz and 64kbps. There were some artifacts on the "aah" sounds, but they don't show up on PC speakers as was suggested. I did struggle with some of the levels, it seems my recording was a little bit on the low side, I was hitting the limiters on my AD261, but may be not often enough! I had to bump it up by 8 to 10dB and compress it to get it approaching a BBC podcast levels, but its my first attempt and I have learned a lot - not so easy getting good sound as they say...

Cheers, Jon