44.1 To 48 Converter
Lida Rick
Ive been using Pro Audio Converter for 10 or so years for work stuff, which involves me batch converting large folders of files. Can set it to replace the source files (which is often what I do) and the file names always stay the same
So I wrote this script, that is executable from Windows, OSX or Linux.
I will put it at the root of the AUDIO directory, on my CF Card.
Once i copied the samples I want > Double click on the script to convert all files found as needed and nothing else to do : D
Cross platform script to convert sampe rate and bit depth of all supported audio files found recursively - GitHub - davidferlay/audio-file-converter: Cross platform script to convert sampe rate and...
Why? When you downsample, you are throwing some data away, since a 48K sample has more data than a 44.1K sample. But upsampling from 44.1K to 48K, you are having to interpolate and actually create data to make up the difference.
Another way you could do is just start a new project in a different folder and run the project setup and set the sample rate to 48k and then import the 44.1k tracks and you will be asked if you want to convert them.
There is one more option: you can select the file then go to: audio>fx>resample. This will also change the sample rate (but probably change your pitch as well).
@garymusic : I think that also works if you remove your file from the project (delete in pool) and just import it again (import or drag/drop), right?
I have several thousand 44.1 Ksps WAV files in a broadcast automation system and I need to convert these files to 48 Ksps. We are switching from using AES3 digital audio to LiveWire AoIP and LiveWire wants 48 Ksps audio. I need to change all these files from one rate to the other without changing their file names. All the files are in one flat directory and my plan would be to write the converted files 2nd directory. The batch converter needs to handle exceptions and not write junk to the new file. It needs to create an error log I can use to resolve problems or at least know the problems are there.
Sadly the program cannot handle exceptions nor create an error logs. Have you try the sample rate the program offers? If you have more specific questions, please open a support ticket through this link: =Switch&support
Hi Putte! Try Barabatch, it does a better job, than the prot@@ls converter...
Ich mach mal auf Deutsch weiter.
Du solltes die Files auf keinen Fall re-recorden, da Du exakten sync brauchst. Beim Aufnehmen knnen leichte Timingverschiebungen auftreten. Du solltest in Deinem Fall Deine Mixdowns auf vollen Sekunden anfangen lassen und dann konvertieren.
Jetzt hast Du's ja bald geschafft ;)
Call Apogee in CA and ask for tech.310 915-1000. Their new PSX100 special edition is killer stuff. Way better than the original. You could probably rent one, or try one out from Guitar center. Not sure this will do what you want, but they're friendly. Good luck.
From my experience the McGill audio tools "resample audio" commandline utility seems to be the best way to do samplerate conversion. It is the defacto standard in the land of audio software development. It supports 24/96kHz.
On an adside, upsampling is never as difficult to do transparently as downsampling because you are not throwing any audio away, so going to 48kHz from 44.1 will be no problem for most software solutions.
My band is working with a somewhat pricey mastering engineer and we are wanting to keep the price down. For a price we're comfortable with, he will either provide us with hi-res masters (24bit, 48khz - the recording quality), or with CD-quality masters + DDP file (16bit-44.1khz).
My question is: what might necessitate a different mastering pass for the CD-quality version? What benefit might there be to getting the mastering engineer to do the CD-quality version instead of just converting it myself from the hi-res version?
Dither, Noise Shaping, and Bit Quantization. These are the reasons for the separate mastering passes. All of which do not need to be considered when mixing/bouncing the audio in the native digital format it was converted to (24bit/48khz).
Also, it sounds like he will provide you with individual "HiRes" native files OR Will downsample for you and create a DDP file. (Which is a FINAL Copy including play order and silence between songs that you can usually upload directly to a CD cuplication service.)
EDIT: I suppose it is possible for the ME to EQ specifically for the lower quality (not likely!) to produce better translation at the format? Also he may handle level peaks differently when downsampling.
For going from 24 bit to 16 bit is relativity simple since it is evenly divisible, however you will still want to apply a dither to make up for the way the originally analog signal loses resolution. This allows simulating a wider set of distinct values than 16 bits would normally allow in exchange for some minimal noise. A great link about this was included by Takuya in the comments below.
With 48khz sampling, you can accurately capture waveforms of up to 24khz frequency. There may be some weird harmonic issues that occur during the conversion, but they would be outside the human hearing range.
I have never considered doing this before but I would like to convert (transcode?) about 140 24 bit high resolution files to 16/44.1 for use with a wireless system (Sonos?) that doesn't support 24 bit. I use XLD and dbPoweramp for Mac for my CD ripping and occasionally transcoding FLAC into AIFF files but I am not aware if either of them can convert 24 to 16 or how you would do it?
In dbPoweramp (Windows) there is a batch conversion capability. Maybe Mac has it too. Regardless, once you pick your files go to "convert" and choose aiff or wav (assuming they are flac to begin with, for example). Then you will get two pulldowns (bit depth, and sample rate). Choose 16 and 44.1. If needed, convert back to FLAC (at 16/44) next.
Other than the NAD Bluesound system it was my understanding that the self contained wireless systems I am looking at (Sonos, Denon HEOS, Polk, Definitive Technology, Bose etc) do not support 24 bit. The NAD is of interest because of its 24/96 support but it is not truly a self contained "wireless" system to me since the speakers require traditional connection and another part of the system requires an ethernet connection.
You could also transcode the files to ALAC, AIFF, or WAV in XLD. For those output formats (but apparently not FLAC), you can click on the "Option" button in Preferences -> General and choose your preferred sample rate and bit depth. It's really a matter of taste/convenience as to whether you use dBpoweramp or XLD.
If you go to ALAC, you can use XLD: In Preferences, choose output format as ALAC, and the Options will let you select 16/44 output. Since you're a mac user, I would recommend this over FLAC - your files will be usable in iTunes directly, and most players will play those downsampled files (except arguably HQPlayer but you have your FLAC or AIFF originals for that!).
XLD usefully checks for corruption in FLAC (by means of checksum embedded in FLAC), and refuse to complete the decoding if corrupt. Some (possibly most) converter apps will blindly convert a corrupt FLAC to another file containing corrupted audio.
Below is a screenshot of XLD detecting corruption in a FLAC. You'll see for yourself by using a hex editor to deliberately change a byte of the audio data in a FLAC file, then trying to convert the corrupted file to some other file with XLD.
A file can be corrupted after its creation; for example, through degradation of storage media. FLAC contains checksums of the contained audio data to help detect such corruption. Some programs, e.g. XLD, make use of the checksums when decoding a FLAC file, and prevent propagation of corruption. Unfortunately, some programs, e.g. Max, do not use the checksums, and they propagate corruption to files which they derive from a corrupt FLAC file.
I had some trouble with minimserver once, with a bunch of AIFF files that did not quite adhere to some (odd) requirement of byte count. I solved this problem by retagging with Yate - basically not changing anything but forcing a file resave. This cleared the issue for thousands of files in my library.
I wonder if your checksum error might be fixed by something like this. Quite possibly the FLAC encoder you used wasn't careful about the checksum - and I wonder if it is possible to fix like I did with Yate.
I switched from USB interface to Dante network and Dante Via only allows sampling rates of 48kHz. Since I've worked in Sonar at 44kHz so far, I now have to change all of my projects. With an audio converter I converted over 1000 wave files from 44kHz to 48kHz in a patch process. In Sonar Preferences for New Projects I have also changed the sampling rate to 48000Hz. If I now load an existing project with the converted wave files, all waves are cut off at the end or set to silence. The file length is retained. If I replace the same file manually via Import audio, the track is displayed and played correctly. So it's not because of the wave file, it is displayed and played correctly in every other player.
What is the difference when Cakewalk loads a project with the associated wave files, or I add the same files manually. In any case, Cakewalk plays the project immediately after opening it, without an error message changed sampling rate. Only the problem that the tracks no longer contain any sound at the end.
in your case, you bought an interface which only supports 48K. so in that case you'll likely have to change the project sample rates on a project by project basis. presumably you're not editing or planning on editing all 200 projects...
The ADAU1442, and the ADAU1452/51/50 will have sample rate converters but it is a complex part to use only as a sample rate converter. If you can use the DSP as well then it may justify the size and cost.
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