[MPlayer-dev-eng] FFmpeg? sound issue

[Ingo Brückl]

Reimar Döffinger wrote on Fri, 18 Nov 2011 18:31:37 +0100:

> On Fri, Nov 18, 2011 at 06:26:55PM +0100, Ingo Brückl wrote:
>> compn wrote on Fri, 18 Nov 2011 11:47:39 -0500:
>>
>> > On Fri, 18 Nov 2011 16:06:37 +0100, Ingo Brückl wrote:
>> >>Today I refreshed my mplayer build (with a new copy of ffmepg - the last one
>> >>was approx. one month old).
>>
>> > could you paste mplayer -v file.mp3 output ?
>>
>> Attached.

> Hm, it outputs float directly to ALSA, maybe ALSA incorrectly thinks
> your soundcard supports that (did you have a recent ALSA/kernel update
> that might have broken it)?

No, I was very inactive recently and only updated wine since my last mplayer
build.

> Try forcing conversion with -af format=s16le

No change.

These are the outputs for:

-afm mp3lib
==========================================================================
Erzwungener Audiocodec: hwac3
Versuche Audiocodecfamilie mp3lib zu erzwingen...
Öffne Audiodecoder: [mp3lib] MPEG layer-2, layer-3
AUDIO: 44100 Hz, 2 ch, s16le, 96.0 kbit/6.80% (ratio: 12000->176400)
Ausgewählter Audiocodec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3)
==========================================================================
AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample)

no option at all
==========================================================================
Erzwungener Audiocodec: hwac3
Öffne Audiodecoder: [ffmpeg] FFmpeg/libavcodec audio decoders
AUDIO: 44100 Hz, 2 ch, floatle, 96.0 kbit/3.40% (ratio: 12000->352800)
Ausgewählter Audiocodec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio)
==========================================================================
AO: [alsa] 44100Hz 2ch floatle (4 bytes per sample)

-af format=s16le
==========================================================================
Erzwungener Audiocodec: hwac3
Öffne Audiodecoder: [ffmpeg] FFmpeg/libavcodec audio decoders
AUDIO: 44100 Hz, 2 ch, floatle, 96.0 kbit/3.40% (ratio: 12000->352800)
Ausgewählter Audiocodec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio)
==========================================================================
AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample)

> And try -ao pcm.

Hmm, that very quickly did play nothing. The dumped file sounds broken as
well.

-ao pcm
==========================================================================
Erzwungener Audiocodec: hwac3
Öffne Audiodecoder: [ffmpeg] FFmpeg/libavcodec audio decoders
AUDIO: 44100 Hz, 2 ch, floatle, 96.0 kbit/3.40% (ratio: 12000->352800)
Ausgewählter Audiocodec: [ffmp3float] afm: ffmpeg (FFmpeg MPEG layer-3 audio)
==========================================================================
[AO PCM] Datei: audiodump.wav (WAVE)
PCM: Samplerate: 44100Hz Kanäle: Stereo Format floatle
[AO PCM] Info: Das Anlegen von Dump-Dateien wird am schnellsten mit
         -benchmark -vc null -vo null -ao pcm:fast erreicht.
[AO PCM] Info: Um WAVE-Dateien zu schreiben, benutze
         -ao pcm:waveheader (Standard).
AO: [pcm] 44100Hz 2ch floatle (4 bytes per sample)

> Bonus points for expanding tests/Makefile to be able to test audio
> while you're at it :-P

I haven't had time yet to look at it and see what this is all about and how
it works.

Ingo