From mboxrd@z Thu Jan 1 00:00:00 1970 From: Jerry Geis Subject: Re: seg fault in libasound Date: Thu, 26 Jun 2008 08:37:15 -0400 Message-ID: <48638D7B.3020207@pagestation.com> References: <486282A1.7050308@pagestation.com> <48638AE3.90003@ladisch.de> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from unifiedpaging.messagenetsystems.com (mail.emergencycommunicationsystems.com [24.123.23.170]) by alsa0.perex.cz (Postfix) with ESMTP id 3AF4F24611 for ; Thu, 26 Jun 2008 14:37:18 +0200 (CEST) In-Reply-To: <48638AE3.90003@ladisch.de> List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: alsa-devel-bounces@alsa-project.org Errors-To: alsa-devel-bounces@alsa-project.org To: Clemens Ladisch Cc: alsa-devel@alsa-project.org List-Id: alsa-devel@alsa-project.org Clemens Ladisch wrote: > Jerry Geis wrote: > >> I am getting a seg fault in alsa. >> >> #2 0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6 >> #3 0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0, >> src_area=0x81cb7ec, src_offset=5223, samples=816, >> format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499 >> > > This is the "assert(bitofs % 8 == 0)" in snd_pcm_channel_area_addr() > in pcm_local.h. In theory, this means that some sample is not aligned > to a byte boundary, but it should be impossible to get this error. > > Do you have any .asoundrc or asound.conf file? If yes, please remove > them and try again. > > > Regards, > Clemens > > You are correct. I removed my /etc/asound.conf file and now it does not seg fault. Below is my file. What is wrong with it? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: default [C-Media USB Headphone Set ], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 lsusb Bus 004 Device 001: ID 0000:0000 Bus 003 Device 002: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter Bus 003 Device 001: ID 0000:0000 Bus 002 Device 002: ID 13fe:1f00 Bus 002 Device 001: ID 0000:0000 Bus 001 Device 001: ID 0000:0000 I have used this file many times with NON USB sound devices and it has been fine. Also - after removing the asound.conf file my sound is a little choppy. Anything I can do about that? Thank you. Jerry ----- # Set default sound card # Useful so that all settings can be changed to a different card here. pcm.snd_card { type hw card 0 device 0 } # Allow mixing of multiple output streams to this device pcm.output { type dmix ipc_key 1024 ipc_perm 0660 # Sound for everybody in your group! slave.pcm "snd_card" slave { # This stuff provides some fixes for latency issues. # buffer_size should be set for your audio chipset. period_time 0 period_size 1024 buffer_size 8192 } bindings { 0 0 1 1 } } # Allow reading from the default device. # Also known as record or capture. pcm.input { type dsnoop ipc_key 2048 slave.pcm "snd_card" ## Possible artsd full duplex fix: # slave { # period_time 0 # period_size 1024 # buffer_size 8192 # } bindings { 0 0 1 1 } } # This is what we want as our default device # a fully duplex (read/write) audio device. pcm.duplex { type asym playback.pcm "output" capture.pcm "input" } ################### # CONVERSION PLUG # ################### # Setting the default pcm device allows the conversion # rate to be selected on the fly. # duplex mode allows any alsa enabled app to read/write # to the dmix plug (Fixes a problem with wine). pcm.!default { type plug slave.pcm "duplex" } # Apparently this is wrong (breaks mplayer for me opening the device) #ctl.!default { # type plug # slave.pcm "snd_card" #} ######## # AOSS # ######## # OSS dsp0 device (OSS needs only output support, duplex will break some stuff) pcm.dsp0 { type plug slave.pcm "output" } # OSS control for dsp0 (needed?...this might not be useful) ctl.dsp0 { type plug slave.pcm "snd_card" } # OSS control for dsp0 (default old OSS is mixer0) ctl.mixer0 { type plug slave.pcm "snd_card" }