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Last post by Niksi -
My music library consists of 1600+ songs, 98% of which are flacs I got from Qobuz, Tidal, and Amazon Music. The other 2% are mp3, opus, m4a. The flacs range from 16 to 24 bits and from 44.1 to 48 kHz. I also used foobar2000 to apply ReplayGain tags, so my volume is normalized.
I want to listen to them using Bluetooth, regardless of its inherit issues with quality audio playback. I don't want to switch to wired headphones, I don't want to use DACs or amps, or anything similar.
My android device currently runs on Android 14, One UI 6.1, and supports LDAC. My headphones (JLab Epic Lab Edition) also support LDAC, and have an EQ that Knowles made specifically for these headphones (which is turned on through the JLab app). I also run an app that forces the LDAC setting in android developer options to always be on "optimized for audio quality" (over 900kbps output).
I understand that LDAC is not lossless, and that I'll never get "bit perfect" over Bluetooth, but is this the best I can get given what I want to use?
I use the Poweramp music player, which utilizes the SoX resampler (on very high quality, 99% cutoff, no dither), I use no Direct Volume Control, and I use AAudio output (which should bypass the default android audio processing).
If anyone has any other suggestions about getting more out of my library, I'm all ears. I'm still pretty new to all this, but I want to learn and enjoy myself.
[...] [aac @ 0000025434729a80] Scalefactor (-6) out of range. [aac @ 0000025434729a80] Scalefactor (257) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (62) exceeds limit (49). [aac @ 0000025434729a80] Scalefactor (-1) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (55) exceeds limit (49). [aac @ 0000025434729a80] Scalefactor (263) out of range. [aac @ 0000025434729a80] Scalefactor (258) out of range. [aac @ 0000025434729a80] Scalefactor (-16) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (60) exceeds limit (49). [aac @ 0000025434729a80] Number of scalefactor bands in group (54) exceeds limit (49). [aac @ 0000025434729a80] Scalefactor (-1) out of range. [aac @ 0000025434729a80] Scalefactor (-1) out of range. [aac @ 0000025434729a80] Scalefactor (-1) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (56) exceeds limit (49). [aac @ 0000025434729a80] Scalefactor (264) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (53) exceeds limit (49). [aac @ 0000025434729a80] Number of scalefactor bands in group (61) exceeds limit (49). [aac @ 0000025434729a80] Number of scalefactor bands in group (15) exceeds limit (14). [aac @ 0000025434729a80] Scalefactor (-2) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (55) exceeds limit (49). [aac @ 0000025434729a80] Number of scalefactor bands in group (50) exceeds limit (49). [aac @ 0000025434729a80] Number of scalefactor bands in group (15) exceeds limit (14). [aac @ 0000025434729a80] Scalefactor (-1) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (60) exceeds limit (49). [aac @ 0000025434729a80] Scalefactor (259) out of range. [aac @ 0000025434729a80] More than one AAC RDB per ADTS frame is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented. [aac @ 0000025434729a80] channel element 1.15 is not allocated [aac @ 0000025434729a80] Number of scalefactor bands in group (15) exceeds limit (14). [aac @ 0000025434729a80] Scalefactor (290) out of range. [aac @ 0000025434729a80] Scalefactor (256) out of range. [aac @ 0000025434729a80] Scalefactor (259) out of range. [aac @ 0000025434729a80] Scalefactor (262) out of range. [aac @ 0000025434729a80] Number of scalefactor bands in group (56) exceeds limit (49). [...]
I will of course look into your feedback. I think it is important to stick to the basic concept of fb2k which is great in my opinion but the execution and the general UX/UI can of course be optimized. Especially the Mac and the mobile versions. It's apparent that no UX/UI person was ever involved in the development process. I still have a Windows PC but I only use it for some occasional gaming these days. Back in the day fb2k was my go to player on Windows. I was also pleased to see that there is a Mac version though it needs some love. ;-) Please ignore the attachments of the first post because they are already outdated and only look into Figma for the current state of affairs. I have no idea if the devs are willing or are capable/have the capacity to adapt any of the suggestions but it's still a fun exercise for me to see how fb2k could be improved. ;-)
Last post by thevudia14 -
I am so sorry about the mixup. you are right. its the same component authored by you. This and crossmix doesn't function together.
Last post by thevudia14 -
i use foo_arg because it eliminates the need to scan the tracks first for rg gain, for any newly added tracks. for some reason, official rg doesn;t work for me, i guess because i dont scan the tracks in my library. foo_arg doesnt ask me to scan anything; i enable it and it works depending on lufs i set it to.
Last post by Case -
I have no idea why you think you need foo_arg. It claims to make RG scanning faster by doing the math incorrectly, but at least on my computer the official RG scanner is much faster. And obviously with foo_arg installed none of the options of RG scanner have any effect since its functionality is replaced by a broken replica. There is nothing for me to fix in relation to this.
Exhale has reached good quality, but is version 1.2.1 the definitive version since it was released in 2023 and ffmpeg usac decoder can already decode sbr correctly?
exhale 1.2.1 will not be the last version, but 1.2.2 most likely will be. I'm planning to release that final version around Christmas.
For those not interested in exhale bitrates higher than ~200 kbps, you can stop reading here. For all others, there's some news.
While working on a small hickup related to SBR preset f (issue #33 for fellow developers), I decided this week to give higher-rate USAC presets a try. First because I found the idea of an archival quality exhale preset compelling for personal use, second because I've seen some threads about new time-domain codecs on HA lately, with claims of "high quality", "full bandwidth", ... whatever. I'm strongly convinced that such codecs are, by design, fundamentally inferior to transform codecs like USAC (though they might be faster, but does that really matter nowadays?).
Anyway, the newest exhale source revision (commit 49517c4d), apart from fixing the preset-f issue and increasing the coding efficiency slightly, includes code which, when activated, allows exhale to encode with up to ~300 kbps (stereo) and up to full CD/broadcast audio bandwidth (24 kHz). To activate this tuning,
change #define SFB_QUANT_PERCEPT_OPT from 1 to 0 in file lib/quantization.h before compiling.
This tuning doesn't make much sense with SBR presets a - g, but with the non-SBR presets, especially 8 and 9, I get nice results.
So, if you're interested in this tuning, please report any compilation errors, audio glitches with these two presets, very high or low rates on specific samples, or anything else which appears fishy. And if someone could provide (or link to) a respective exhale executable compiled for others to do the tests, that would be appreciated as well.
Thanks!
My analysis results for stereo, presets > 4 and different sampling rates:
- for 44.1 kHz sampling rate: avg bitrate = preset * 32 kbps - for >44 kHz sampling rate: avg rate proportionally higher - for 32.0 kHz sampling rate: 16 kbps less than for 44.1 kHz
Developer note (mainly for personal reference, in case somebody asks):
- to get decoded waveforms identical to those created by exhale 1.2.1, set #define SFB_QUANT_PERCEPT_OPT to 2.