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Thanks for the bug report. Main problem is that "no DSP reset" mode treats all input as one audio stream and partitions output back into tracks, making use of original track lengths + reported DSP latency. After something failed in the middle, the output no longer really makes sense and the conversion goes off the rails, precisely as you've observed.
Ok thanks. I now understand why it all goes pear-shaped.
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I think you want to convert each album as a separate batch.
That makes sense, except this was a fresh transcode of my entire library, so doing it all in one go was much easier. The reason for the fresh transcode was because previously (without the "no DSP reset") I had too many instances of little glitches at the boundaries of tracks that ran into one another (eg. things like Dark Side Of The Moon, Beethoven Symphony 5, etc). My library is nearly all 44.1kHz, with just a small smattering of 48kHz albums (typically ripped from DVDs).
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Also, why not just resample everything to a fixed rate before doing any other DSP?
Yes, good idea. I frankly didn't even remember I had some 48kHz albums until I Googled what the error message actually meant.
So, how 18kbps and 20kbps files are CBR and they are working with Flash Player? I'm uploading a 20kbps file example...
What makes you think it's 20 kbps CBR? Because some player told you so? Have you ever thought of double-checking this value via others apps? Especially via apps that are designed to analyze media files? It is not a CBR file, but a VBR (or ABR) file without a proper header, which data is encoded using 16 kbps and 24 kbps, so it gives an average of 20 kbps, since (16×9+24×9)/18=20.
Audio Format : MPEG Audio Format version : Version 2.5 Format profile : Layer 3 Bit rate mode : Variable Channel(s) : 1 channel Sampling rate : 11.025 kHz Compression mode : Lossy Stream size : 2.29 KiB (100%)
$ mp3val.exe google.mp3
WARNING: VBR detected, but no VBR header is present. Seeking may not work properly. WARNING: No supported tags in the file INFO: 18 MPEG frames (MPEG 2.5 Layer III), no tags, no VBR header
Last post by KevinB52379 -
I tried that build and I get about 300x from the external hard drive where my FLACs are and converting to mp3 to the internal SSD using foobar2000 and Helix.
I am copying the FLACs over to the internal SSD and I'm going to try using foobar2000 to convert totally from internal SSD drive...so the source will be the internal SSD and the destination will be the same internal SSD.
My system is a PCIe Gen 3x so I didn't bother paying more for a PCIE gen 4 drive.
There were no BIOS updates. The latest is from October 2023 and that is what I have installed.
Last post by Wombat -
Here is an x64 compile that is built with the slower compiler option -O1 that is only minimal faster as the win32 by Case. The idea is that something overpaces your config. Also a test with files encoded from the local SSD without USB in play can help to get an idea.
Last post by KevinB52379 -
I'll look for a BIOS update. The one I currently have installed is from October 2023. It's an HP system. But, I feel that if it was a USB problem, wouldn't it be happening with Case's version as well?
I'll search on HP's website for a newer BIOS offering for my model.
Last post by Wombat -
I still wonder what happens here and the following idea may be far fetched. Together with my B550 chipset for AMD i had several USB problems until some AGESA BIOS update fixed it. It may be worth to check for the latest BIOS update.
Last post by Klymins - @AiZ How can i get this with ~12kbps (ABR does not do that)? And, which program is shown in this screenshot? Addition: By "ABR does not do that", i mean: Let 8kbps frames be letter A and 16kbps ones B. ABR does something like "ABABBAABBABAABABBA" but i want "BABABABABABABABA".
These are the bitrates for MP3 CBR mode. There is no 12 kbps among them. Thus, to get 12 kbps file, which can be played back on a wide range of audio equipment, you have no choice but to use VBR mode (or ABR as a variation of it) so that some of the data is compressed at a lower bitrate and some of the data is compressed at a higher bitrate, producing the output that is as close as possible to the bitrate you specified.
If not specified, LAME may sometimes resample automatically when faced with extreme compression conditions (like encoding a 44.1 kHz input file at 32 kbps). To disable this automatic resampling, you have to use --resample to set the output samplerate equal to the input samplerate. In that case, LAME will not perform any extra computations.