Tested few common audio formats by phase cancellation method to grade the quality of audio compressions. It is known that the psychoacoustics model of compression does not allow to conclude the quality of encode by the quantity of data thrown out rather by what quality of data is thrown out. Theoretically a 128 Kbps mp3 file recoded to 160 kpbs should not lose any data for compression, but practically that does not happen.
So I feel when it comes to judging the quality of sound, eyes are more better than ears for inference! ABX listening tests may not be very reliable.
The method of phase cancellation method is well known yet am mentioning it for ready reference. Everybody can perform it in their own system. In a nutshell I will say the encoded waveform should be mixed with the inverse of the actual waveform to know the difference of data left out. Here are the details
In adobe audition multitrack view (or any other sound editor) when the source and encoded are viewed there is a small sync difference. For proper mixing I introduced manually a crust in waveform as an anchor point in the very beginning of the file.
The waveform are now synched by cutting a small portion towards the left of the encoded waveform and moving the crust to match the position of the original waveform. In the fourth image, the vertical bar shows the position the cut was made in the encoded waveform.
Having inverted the source in single track view (Edit View) by Effects->Invert and return to multitrack view Edit-Mixdown to file (all audio clips) is all one needs to do to find out the difference of data thrown out
As an alternative one can have the copy of INVERTED source file in the first track and original source file in the third track and save the document. The encoded file will be placed in the second track and after synching the audio with the third track source as mentioned above, the actual source can be deleted and the mix made and screenshot taken and the audio document reverted for placing the other encoded file in the second track(if necessary). For perfect encodes like wma, ogg, lame mp3 the original file can be opened and edit->mixpaste with invert of the encoded file as shown in the fifth image.
I did not wish to encode any file below 128 Kbps because even good encoders performed only marginally better compared to 128 bit mp3 files. With AAC HE high efficiency mode I could not match the crust of encoded file and hence did not consider it probably they dont compress well.
I feel helix mp3 at 320 is the best codec considering the quality of encode and ubiquitous nature of mp3 .
CBR is always better than VBR
The images that you are viewing is the cropped image of left channel (the right being similar is not considered) stacked one above other for easy inference. The images are almost in alignment with other images and can be visually compared to-and-fro in image viewer. Unable to upload lot of files and hence attaching a zipped version.
Software used
1. Adobe audition
2. EZ Audio converter
3. Mediacoder
4. Surcode Standalone encoder
5. DTS HD Master and
6. Arcsoft DTS decoder
So I feel when it comes to judging the quality of sound, eyes are more better than ears for inference! ABX listening tests may not be very reliable.
The method of phase cancellation method is well known yet am mentioning it for ready reference. Everybody can perform it in their own system. In a nutshell I will say the encoded waveform should be mixed with the inverse of the actual waveform to know the difference of data left out. Here are the details
In adobe audition multitrack view (or any other sound editor) when the source and encoded are viewed there is a small sync difference. For proper mixing I introduced manually a crust in waveform as an anchor point in the very beginning of the file.
The waveform are now synched by cutting a small portion towards the left of the encoded waveform and moving the crust to match the position of the original waveform. In the fourth image, the vertical bar shows the position the cut was made in the encoded waveform.
Having inverted the source in single track view (Edit View) by Effects->Invert and return to multitrack view Edit-Mixdown to file (all audio clips) is all one needs to do to find out the difference of data thrown out
As an alternative one can have the copy of INVERTED source file in the first track and original source file in the third track and save the document. The encoded file will be placed in the second track and after synching the audio with the third track source as mentioned above, the actual source can be deleted and the mix made and screenshot taken and the audio document reverted for placing the other encoded file in the second track(if necessary). For perfect encodes like wma, ogg, lame mp3 the original file can be opened and edit->mixpaste with invert of the encoded file as shown in the fifth image.
I did not wish to encode any file below 128 Kbps because even good encoders performed only marginally better compared to 128 bit mp3 files. With AAC HE high efficiency mode I could not match the crust of encoded file and hence did not consider it probably they dont compress well.
I feel helix mp3 at 320 is the best codec considering the quality of encode and ubiquitous nature of mp3 .
CBR is always better than VBR
The images that you are viewing is the cropped image of left channel (the right being similar is not considered) stacked one above other for easy inference. The images are almost in alignment with other images and can be visually compared to-and-fro in image viewer. Unable to upload lot of files and hence attaching a zipped version.
Software used
1. Adobe audition
2. EZ Audio converter
3. Mediacoder
4. Surcode Standalone encoder
5. DTS HD Master and
6. Arcsoft DTS decoder