Setting one will set the other as well, since you can't have prediction without the profile being set and you can't have the profile set without prediction (well you can but it would be a hack as you'd just set all scalefactor bands to disable prediction). I couldn't make the changing bandwidth work in a stable fashion without a lot more work, so I'll reserve that for a further patch, maybe. I think it's not specific to this patch, but I just noticed it. Asking for help, clarification, or responding to other answers. It was a very simple thing, but very hard to find. You seem to have either a bad patch or a misapplied patch, because that's the bug I solved, and in my build it works fine. Thank you. Improved VBR, fixed psy threshold reduction bug. Next post ought to be a patch. By clicking “Post Your Answer”, you agree to our terms of service, privacy policy and cookie policy. It happens at higher bitrates. I think it will contribute to higher quality in 160kbps and 192kbps. http://www.ffmpeg.org/releases/ffmpeg-1.2.1.tar.bz2. in aaccoder.c twoloop line 795, and found cutoff differs between different frames. A way to squeeze a little extra bandwidth. You can also try adding the Motumedia PPA to your apt sources and update your ffmpeg packages. FFmpeg_anmr_error7.flac still stops FFmpeg on options -aac_coder faac and fast. The bug is most obvious at -b:a 224k. Some bug I couldn't find with encode_window_bands_info, but I just made anmr use the other (codebook_trellis_rate? You could try confirming whether earlier revisions also exhibit that behavior, and how far back. I am no audiophile and a little hard of hearing so I cannot find fault with the Native Encoder but I can tell the difference between 2 and 6 channels :-). However, still, it has a distinctive bug. The sound disappears partially when the sound is white noise-like. The output using ffmpeg -i input.wav -vn -ac 2 -b:a 160k output1.mp3 is constant bit rate, however ffmpeg writes a header with the title Xing and Mediainfo infers that to indicate VBR. Thank you. Comparison between v6 patch and v7 patch. Thank you, but it still crashes on 80kbps. I tried Nero abr, but I could not set it to 96kbps. Alright, attaching another version. So I re-did twoloop's RD improvement step to better respect hole avoidance, to be asymmetric in its scale manipulation (ie: to avoid adding all 1 or all 2, which would be quickly undone by the bitrate adjustment step), and everything seemed to work a lot better. was investigated again by looking into full behavior of static void encode_scale_factors(). I've got a request of testing libfaac, mp2, and eac3, but I'm running out of the "slot". klaussfreire, could you provide the VBR-only patch? I worked hard to get good results, but there's still problematic samples, that sound better on equivalent ABR than VBR. Can you share a sample that reproduces the crash? Average Speed: 13.6x -cutoff 18000 seems to work, but the lowpass filter is too dull, compared to many practical encoders. `-codec:a libfaac` always encodes with a bitrate between 150k and 160k when `-b:a` is above 160k. No, the current git master contains no changes from the previous v9b or the future v9c yet. I will tomorrow. "AAC: Add support for 7350Hz sampling rates, no error on too hight bitrate.". Craptastic. The patch contains trailing white space, this cannot be pushed to our git repository, please remove it. It's not commitable or complete yet. aacenc.c line 106 and 133, this properly worked, and the result 7350Hz aac mp4s were playable on FFmpeg, foobar2000 v1.2.9 and Media Player Classic. Min / Average / Max bitrate: 130 / 134 / 141 kbps If the patch improves VBR but causes issues for CBR, thats bad. No, the cutoff moves up and down, but the LP remains on. If not, I think I should test v4 ABR first, because it's stable, have less artifacts in tonal samples. Average Speed: 11.1x If you want to preserve higher frequencies, use -cutoff 18000. Later today, or perhaps tomorrow, depending on your time zone.
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