Je suis sous Archlinux 64bit, comme je développe un peut j'ai décidé d'utiliser la library openAL pour gérer le son, mais aucun son n'est émis.
Le problème ne vient pas de mon code car il fonctionne très bien sur mon netbook (lui même sous Archlinux mais en 32bit) avec le même programme, la même musique ...
Je suppose donc que cela vient d'un problème de configuration de openAL, mais comme je connais assez mal cette library j'ai besoin de votre aide.
Voici mon /etc/openal/alsoft.conf :
Code : Tout sélectionner
# OpenAL config file. Options that are not under a block or are under the
# [general] block are for general, non-backend-specific options. Blocks may
# appear multiple times, and duplicated options will take the last value
# specified.
# The system-wide settings can be put in /etc/openal/alsoft.conf and user-
# specific override settings in ~/.alsoftrc.
# For Windows, these settings should go into %AppData%\alsoft.ini
# The environment variable ALSOFT_CONF can be used to specify another config
# override
# Option and block names are case-insenstive. The supplied values are only
# hints and may not be honored (though generally it'll try to get as close as
# possible). Note: options that are left unset may default to app- or system-
# specified values. These are the current available settings:
## format:
# Sets the output format. Can be one of:
# AL_FORMAT_MONO8 (8-bit mono)
# AL_FORMAT_STEREO8 (8-bit stereo)
# AL_FORMAT_QUAD8 (8-bit 4-channel)
# AL_FORMAT_51CHN8 (8-bit 5.1 output)
# AL_FORMAT_61CHN8 (8-bit 6.1 output)
# AL_FORMAT_71CHN8 (8-bit 7.1 output)
# AL_FORMAT_MONO16 (16-bit mono)
# AL_FORMAT_STEREO16 (16-bit stereo)
# AL_FORMAT_QUAD16 (16-bit 4-channel)
# AL_FORMAT_51CHN16 (16-bit 5.1 output)
# AL_FORMAT_61CHN16 (16-bit 6.1 output)
# AL_FORMAT_71CHN16 (16-bit 7.1 output)
# AL_FORMAT_MONO32 (32-bit float mono)
# AL_FORMAT_STEREO32 (32-bit float stereo)
# AL_FORMAT_QUAD32 (32-bit float 4-channel)
# AL_FORMAT_51CHN32 (32-bit float 5.1 output)
# AL_FORMAT_61CHN32 (32-bit float 6.1 output)
# AL_FORMAT_71CHN32 (32-bit float 7.1 output)
format = AL_FORMAT_STEREO16
## cf_level:
# Sets the crossfeed level for stereo output. Valid values are:
# 0 - No crossfeed
# 1 - Low crossfeed
# 2 - Middle crossfeed
# 3 - High crossfeed (virtual speakers are closer to itself)
# 4 - Low easy crossfeed
# 5 - Middle easy crossfeed
# 6 - High easy crossfeed
# Users of headphones may want to try various settings. Has no effect on non-
# stereo modes.
cf_level = 0
## head_dampen:
# Sets the amount of dampening on sounds emanating from behind the listener.
# This is used to simulate the natural occlusion of the head, which is
# typically missing with mono and stereo output, and as such, only works on
# mono and stereo output modes. Valid values range from 0 to 1 (inclusive),
# and higher values provide a stronger effect.
head_dampen = 0.25
## frequency:
# Sets the output frequency.
frequency = 44100
## resampler:
# Selects the resampler used when mixing sources. Valid values are:
# 0 - None (nearest sample, no interpolation)
# 1 - Linear (extrapolates samples using a linear slope between samples)
# 2 - Cubic (extrapolates samples using a Catmull-Rom spline)
# Specifying other values will result in using the default (linear).
resampler = 1
## rt-prio:
# Sets real-time priority for the mixing thread. Not all drivers may use this
# (eg. PortAudio) as they already control the priority of the mixing thread.
# 0 and negative values will disable it. Note that this may constitute a
# security risk since a real-time priority thread can indefinitely block
# normal-priority threads if it fails to wait. As such, the default is
# disabled.
rt-prio = 0
## period_size:
# Sets the update period size, in frames. This is the number of frames needed
# for each mixing update.
period_size = 1024
## periods:
# Sets the number of update periods. Higher values create a larger mix ahead,
# which helps protect against skips when the CPU is under load, but increases
# the delay between a sound getting mixed and being heard.
periods = 4
## sources:
# Sets the maximum number of allocatable sources. Lower values may help for
# systems with apps that try to play more sounds than the CPU can handle.
sources = 256
## stereodup:
# Sets whether to duplicate stereo sounds on the rear and side speakers for 4+
# channel output. This provides a "fuller" playback quality for 4+ channel
# output modes, although each individual speaker will have a slight reduction
# in volume to compensate for the extra output speakers. True, yes, on, and
# non-0 values will duplicate stereo sources. 0 and anything else will cause
# stereo sounds to only play out the front speakers. This only has an effect
# when a suitable output format is used (ie. those that contain side and/or
# rear speakers).
stereodup = true
## scalemix:
# Sets whether to scale the remixed output. When the final mix is written to
# the device, the multi-channel data is remixed so pure-virtual channels (eg.
# front-center on stereo output) are remixed and added to available channels
# (eg. front-left and front-right). Scaling helps ensure that no single source
# will put out more than 100% on a given physical channel. This can cause a
# noticeable reduction in overall volume, however, so it is off by default.
scalemix = false
## drivers:
# Sets the backend driver list order, comma-seperated. Unknown backends and
# duplicated names are ignored. Unlisted backends won't be considered for use
# unless the list is ended with a comma (eg. 'oss,' will list OSS first
# followed by all other available backends, while 'oss' will list OSS only).
# Backends prepended with - won't be available for use (eg. '-oss,' will allow
# all available backends except OSS). An empty list means the default.
# drivers = pulse,alsa,oss,solaris,dsound,winmm,port,null,wave
drivers = alsa
## excludefx:
# Sets which effects to exclude, preventing apps from using them. This can
# help for apps that try to use effects which are too CPU intensive for the
# system to handle. Available effects are: eaxreverb,reverb,echo,modulator
excludefx =
## slots:
# Sets the maximum number of Auxiliary Effect Slots an app can create. A slot
# can use a non-negligible amount of CPU time if an effect is set on it even
# if no sources are feeding it, so this may help when apps use more than the
# system can handle.
slots = 4
## sends:
# Sets the number of auxiliary sends per source. When not specified (default),
# it allows the app to request how many it wants. The maximum value currently
# possible is 4.
sends =
## layout:
# Sets the virtual speaker layout. Values are specified in degrees, where 0 is
# straight in front, negative goes left, and positive goes right. Unspecified
# speakers will remain at their default positions (which are dependant on the
# output format). Available speakers are back-left(bl), side-left(sl), front-
# left(fl), front-center(fc), front-right(fr), side-right(sr), back-right(br),
# and back-center(bc).
layout =
## layout_*:
# Channel-specific layouts may be specified to override the layout option. The
# same speakers as the layout option are available, and the default settings
# are shown below.
#layout_STEREO = fl=-90, fr=90
#layout_QUAD = fl=-45, fr=45, bl=-135, br=135
#layout_51CHN = fl=-30, fr=30, fc=0, bl=-110, br=110
#layout_61CHN = fl=-30, fr=30, fc=0, sl=-90, sr=90, bc=180
#layout_71CHN = fl=-30, fr=30, fc=0, sl=-90, sr=90, bl=-150, br=150
##
## ALSA backend stuff
##
[alsa]
## device:
# Sets the device name for the default playback device.
device = default
## capture:
# Sets the device name for the default capture device.
capture = default
## mmap:
# Sets whether to try using mmap mode (helps reduce latencies and CPU
# consumption). If mmap isn't available, it will automatically fall back to
# non-mmap mode. True, yes, on, and non-0 values will attempt to use mmap. 0
# and anything else will force mmap off.
mmap = true
##
## OSS backend stuff
##
[oss]
## device:
# Sets the device name for OSS output.
device = /dev/dsp
## capture:
# Sets the device name for OSS capture.
capture = /dev/dsp
##
## Solaris backend stuff
##
[solaris]
## device:
# Sets the device name for Solaris output.
device = /dev/audio
##
## DirectSound backend stuff
##
[dsound]
##
## Windows Multimedia backend stuff
##
[winmm]
##
## PortAudio backend stuff
##
[port]
## device:
# Sets the device index for output. Negative values will use the default as
# given by PortAudio itself.
device = -1
## capture:
# Sets the device index for capture. Negative values will use the default as
# given by PortAudio itself.
capture = -1
##
## PulseAudio backend stuff
##
[pulse]
## spawn-server:
# Attempts to spawn a PulseAudio server when requesting to open a PulseAudio
# device. Note that some apps may open and probe all enumerated devices on
# startup, causing a server to spawn even if a PulseAudio device is not
# actually selected. Setting autospawn to false in Pulse's client.conf will
# still prevent autospawning even if this is set to true.
spawn-server = false
##
## Wave File Writer stuff
##
[wave]
## file:
# Sets the filename of the wave file to write to. An empty name prevents the
# backend from opening, even when explicitly requested.
# THIS WILL OVERWRITE EXISTING FILES WITHOUT QUESTION!
file =Merci d'avance pour vos futures réponsesAvailable playback devices:
ALSA Default
HDA Intel PCH [ALC892 Analog] (hw:0,0) via ALSA
HDA Intel PCH [ALC892 Digital] (hw:0,1) via ALSA
HD-Audio Generic [HDMI 0] (hw:1,3) via ALSA
Available capture devices:
ALSA Default
HDA Intel PCH [ALC892 Analog] (hw:0,0) via ALSA
Default playback device: ALSA Default
Default capture device: ALSA Default
ALC version: 1.1
** Info for device "ALSA Default" **
ALC version: 1.1
ALC extensions:
ALC_ENUMERATE_ALL_EXT, ALC_ENUMERATION_EXT, ALC_EXT_CAPTURE,
ALC_EXT_disconnect, ALC_EXT_EFX, ALC_EXT_thread_local_context
OpenAL vendor string: OpenAL Community
OpenAL renderer string: OpenAL Soft
OpenAL version string: 1.1 ALSOFT 1.13
OpenAL extensions:
AL_EXT_DOUBLE, AL_EXT_EXPONENT_DISTANCE, AL_EXT_FLOAT32, AL_EXT_IMA4,
AL_EXT_LINEAR_DISTANCE, AL_EXT_MCFORMATS, AL_EXT_MULAW,
AL_EXT_MULAW_MCFORMATS, AL_EXT_OFFSET, AL_EXT_source_distance_model,
AL_LOKI_quadriphonic, AL_SOFT_buffer_sub_data, AL_SOFT_loop_points
EFX version: 1.0
Max auxiliary sends: 1
Supported filters:
Low-pass
Supported effects:
EAX Reverb, Reverb, Echo, Ring Modulator