[prev in list] [next in list] [prev in thread] [next in thread] 

List:       amarok-bugs-dist
Subject:    [amarok] [Bug 350724] New: Limits playback to 48KHz
From:       Aaron Williams <aaronw () doofus ! org>
Date:       2015-07-28 22:25:43
Message-ID: bug-350724-71684 () http ! bugs ! kde ! org/
[Download RAW message or body]

https://bugs.kde.org/show_bug.cgi?id=350724

            Bug ID: 350724
           Summary: Limits playback to 48KHz
           Product: amarok
           Version: 2.8.0
          Platform: Other
                OS: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: NOR
         Component: Playback
          Assignee: amarok-bugs-dist@kde.org
          Reporter: aaronw@doofus.org
                CC: matej@laitl.cz

I have a number of FLAC files that are encoded with 24-bit, 96KHz or higher and
playback is always limited to 48KHz, 24 bit according to ALSA. I have
configured ALSA's DMIX to use 96KHz but Amarok still only uses 48KHz.

Reproducible: Always

Steps to Reproduce:
1. Configure Alsa DMIX for 96KHz
In /usr/share/alsa/alsa.conf:
defaults.pcm.dmix.rate 96000 
2. Play a 96KHz FLAC file in Amarok
3. Look at /proc/asound/card0/stream0

Here is what /proc/asound/card0/stream0 shows when playing a 96KHz, 24-bit FLAC
file in Amarok:
GFEC ASSP DigiHug USB Audio at usb-0000:00:14.0-2, full speed : USB Audio

Playback:
  Status: Running
    Interface = 3
    Altset = 1
    Packet Size = 243
    Momentary freq = 48000 Hz (0x30.0000)
  Interface 3
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 3 OUT (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000
  Interface 3
    Altset 2
    Format: S24_3LE
    Channels: 2
    Endpoint: 3 OUT (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000

Capture:
  Status: Stop
  Interface 2
    Altset 1
    Format: S16_LE
    Channels: 2
    Endpoint: 2 IN (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000
  Interface 2
    Altset 2
    Format: S24_3LE
    Channels: 2
    Endpoint: 2 IN (ADAPTIVE)
    Rates: 8000, 16000, 32000, 44100, 48000, 96000

I am hearing a noticeable difference in the sound quality when the sound is
down-mixed to a lower frequency. This is especially apparent in one recording I
have where I have two encodings, 44100 at 20-bits (after HDCD decode) and at
176400, 24-bits. The 44100 encoding sounds noticeably better and the only thing
I can conclude is that it's due to the samplerate conversion from 176400 to
48000 in ALSA. I imagine it would be better if it were instead converted to
96000.


Actual Results:  
Output is downsampled to 48KHz.

Expected Results:  
Output should be at 96KHz.

This is using an Aune T1 Mark 2 24-bit, 96KHz USB tube DAC.

-- 
You are receiving this mail because:
You are the assignee for the bug.
_______________________________________________
Amarok-bugs-dist mailing list
Amarok-bugs-dist@kde.org
https://mail.kde.org/mailman/listinfo/amarok-bugs-dist

[prev in list] [next in list] [prev in thread] [next in thread]