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Re: Can't play 24bit/96kHz flac files





On Sat, May 1, 2010 at 12:27 PM, Anand Sivaram <aspnair@gmail.com> wrote:
On Sat, May 1, 2010 at 15:43, James Stuckey <jhstuckey@gmail.com> wrote:
> If I try to play them in MOC I get: (0) Can't set audio parameters: Invalid
> argument
>
> With mplayer I see:
>
> MPlayer SVN-r30656 (C) 2000-2010 MPlayer Team
> Can't open joystick device /dev/input/js0: No such file or directory
> Can't init input joystick
> mplayer: could not connect to socket
> mplayer: No such file or directory
> Failed to open LIRC support. You will not be able to use your remote
> control.
>
> Playing 02 Lohengrin - Prelude, Act 1.flac.
> Audio only file format detected.
> ==========================================================================
> Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
> AUDIO: 96000 Hz, 6 ch, s32le, 6195.6 kbit/33.61% (ratio: 774455->2304000)
> Selected audio codec: [ffflac] afm: ffmpeg (FFmpeg FLAC audio)
> ==========================================================================
> [AO_ALSA] alsa-lib: pcm_hw.c:1293:(snd_pcm_hw_open) open /dev/snd/pcmC0D0p
> failed: Device or resource busy
> [AO_ALSA] Playback open error: Device or resource busy
> Failed to initialize audio driver 'alsa'
> [AO OSS] audio_setup: Can't open audio device /dev/dsp: Device or resource
> busy
> [AO_ALSA] alsa-lib: pcm_hw.c:1293:(snd_pcm_hw_open) open /dev/snd/pcmC0D0p
> failed: Device or resource busy
> [AO_ALSA] Playback open error: Device or resource busy
> [AO ARTS] loading the aRts backend "/usr/lib/libartscbackend.la" failed
> [AO ESD] esd_open_sound failed: Connection timed out
> AO: [pulse] Init failed: Connection refused
> [JACK] cannot open server
> ao_nas: init(): Can't open nas audio server -> nosound
> [AO SDL] Samplerate: 96000Hz Channels: Stereo Format s32le
> [AO SDL] Unsupported audio format: 0x19.
> [AO_ALSA] alsa-lib: pcm_hw.c:1293:(snd_pcm_hw_open) open /dev/snd/pcmC0D0p
> failed: Device or resource busy
> [AO SDL] Unable to open audio: Couldn't open audio device: Device or
> resource busy
> DVB card number must be between 1 and 4
> AO: [null] 96000Hz 6ch s32le (4 bytes per sample)
> Video: no video
> Starting playback...
> A:   3.5 (03.5) of 570.0 (09:30.0)  1.4%
> Exiting... (Quit)
>
> Running file on the file gives:
> flac: FLAC audio bitstream data, 24 bit, 6 channels, 96 kHz, 54720000
> samples
>
> This only happens with this file. The file plays on other systems (windows).
>

Most of the cheap in built audio cards could only support 16bit 44100Hz/48000Hz
So just see the specification of your audio card.
But audio players could down sample them to 16bit 44100Hz/48000Hz,
windows player may be doing that.  But of course, that reduces the
quality.


Ah, okay. So the decoding is being done on the sound card? In what sense does the sound card need to support the file type?

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