Playback limited to s16le?

[idl0r]

Hi,

base on #3091 I seem to be able to let it upscale to 196kHz but it keeps sticking to s16le even though float32le has been set as default for pulseaudio. I was looking at it using "pulseeffects" which shows the actual information. So to me it looks like s16le is max/hardcoded or something.

http://www.2l.no/hires/ http://www.lindberg.no/hires/test/2L-145/2L-45_stereo_01_FLAC_176k_24b.flac % file 2L-45_stereo_01_FLAC_176k_24b.flac 2L-45_stereo_01_FLAC_176k_24b.flac: FLAC audio bitstream data, 24 bit, stereo, 176.4 kHz, 54206544 samples

Playing with mplayer works just fine. It even seems to upsample 16bit stuff.

Also, my settings are on "Choose automatically" and it still doesn't work. It also doesn't work when I select anything else in that list.

In my case it's clementine 1.3.1 based on 9af827b6acaabb2331246f58436cd34b11548b6c

[v-fox]

What exactly you think is limited ? Seems like you've made some wrong assessments in what encodes what and how:

• Sample frequency, by definition, is not tied to encoding format.
• Settings for frequency and format in PulseAudio are intended for sinks and sources, meaning - hardware, unless you use PA as JACK's compatibility shim, which uses float32, or some PA filter-sinks. Otherwise you should use what your hardware advertises, such as /proc/asound//stream0 for snd-usb-audio and /proc/asound//codec#0 for HDA. I'm yet to see any actual hardware with native float encoding. Audio pipeline would either fail or silently downsample to whatever it considers a "safe fallback" on impossible demands.
• Then you configured your hardware, you can start configuring intermediate software processing to be 96-192khz/float32 but your target still has to be what hardware expects. Use paman, pavucontrol and pasystray to see what PA actually does and configure it on the fly. Some heavy processing stages with sample-rates higher than 48k may crush your CPU when done in realtime and with small buffering to decrease latency.

I use native JACK GST backend in Clementine, so it bypasses PA altogether. It seem to use default GST logic for upsampling, as I complained about in https://github.com/clementine-player/Clementine/issues/6132

[idl0r]

Well, "file" says it's 24bit flac, mplayer plays it that way and pulseaudio/pulseeffects shows it that way. When I play that via clementine it always sticks to s16le, not higher, even though pulse is configured to a higher option.

[jonaski]

@idl0r is right. The bit depth is limited because the caps set for the probe is also affecting the other caps (audio queue / converter). The analyzer needs signed 16 bit (S16LE) to work. We use a tee element to split the pipeline. The splitting is supposed work so that the audio queue and converter is unaffected, but this isn't working, and have never worked. After commit https://github.com/clementine-player/Clementine/commit/1c0891202d582b8a7ca96f73e517b24120b6ac09#diff-a1f8da8007e51dcea858c84fe1eedba2 it definitely won't work, because the probe queue is forced to 16 bit, but the audio queue is set to be auto negotiated, that means that the both will go to 16 bit anyway which means the whole splitting is a waste. For it to work you must force the format on the other caps too. I've tried that too, and had no luck, I can get it to work using gst-launch-1.0, but not in code. If I had the solution I would have proposed a change already. Even though it shows up to be playing at S24LE, it's not actually, the buffer is already converted, I see it because I have a DAC that shows bit depth. Even though some people will probably think they are playing 24 bit (or whatever), they are are actually fooled by the capabilities of their DAC. The relevant code is here: https://github.com/clementine-player/Clementine/blob/master/src/engines/gstenginepipeline.cpp#L441 In my fork I have removed the limiting, but it has the disadvantage that the analyzer doesn't work for all formats.

[v-fox]

@jonaski Damn, I can't believe they would screw the pooch that hard ! …although, after years of being qt4-only, tied to deprecated, disabled by default, sqlite FTS function and transcoding garbling up AAC, I should.

Does this happen even with JACK as output, it downconverts then upconverts ? Anyway, I guess it's time to check out the fork. As long as it has moodbar and JACK output.

[jonaski]

Yes it happens with all formats and any output.