Comments (9)
Hello,
can you tell us your baresip version and paste your ~/.baresip/config please.
Thank you!
from baresip.
I have this problem with version 0.4.14
and with the GitHub version. For both builds I use re-0.4.13
and rem-0.4.6
.
Here is my ~/.baresip/config
:
#
# baresip configuration
#
#------------------------------------------------------------------------------
# Core
poll_method epoll # poll, select, epoll ..
# SIP
sip_trans_bsize 128
#sip_listen 0.0.0.0:5060
#sip_certificate cert.pem
# Audio
audio_player alsa,default
audio_source alsa,default
audio_alert alsa,default
audio_srate 8000-48000
audio_channels 1-2
#ausrc_srate 48000
#auplay_srate 48000
#ausrc_channels 0
#auplay_channels 0
# Video
#video_source v4l2,/dev/video0
#video_display x11,nil
video_size 352x288
video_bitrate 500000
video_fps 25
# AVT - Audio/Video Transport
rtp_tos 184
#rtp_ports 10000-20000
#rtp_bandwidth 512-1024 # [kbit/s]
rtcp_enable no
rtcp_mux no
jitter_buffer_delay 5-10 # frames
rtp_stats no
# Network
#dns_server 10.0.0.1:53
#net_interface wlan0
# BFCP
#bfcp_proto udp
#------------------------------------------------------------------------------
# Modules
#module_path /usr/local/lib/baresip/modules
# UI Modules
module stdio.so
#module cons.so
#module evdev.so
#module httpd.so
# Audio codec Modules (in order)
#module opus.so
#module silk.so
#module amr.so
#module g7221.so
#module g722.so
#module g726.so
module g711.so
#module gsm.so
#module l16.so
#module speex.so
#module bv32.so
# Audio filter Modules (in encoding order)
module vumeter.so
#module sndfile.so
#module speex_aec.so
#module speex_pp.so
#module plc.so
# Audio driver Modules
module alsa.so
#module portaudio.so
# Video codec Modules (in order)
#module avcodec.so
#module vpx.so
# Video filter Modules (in encoding order)
#module selfview.so
# Video source modules
#module v4l.so
#module v4l2.so
#module x11grab.so
#module cairo.so
# Video display modules
#module x11.so
#module sdl2.so
# Audio/Video source modules
#module rst.so
#module gst.so
# Media NAT modules
module stun.so
module turn.so
module ice.so
module natpmp.so
# Media encryption modules
#module srtp.so
#module dtls_srtp.so
#------------------------------------------------------------------------------
# Temporary Modules (loaded then unloaded)
module_tmp uuid.so
module_tmp account.so
#------------------------------------------------------------------------------
# Application Modules
module_app auloop.so
module_app contact.so
module_app menu.so
#module_app mwi.so
#module_app natbd.so
#module_app presence.so
#module_app syslog.so
module_app vidloop.so
#------------------------------------------------------------------------------
# Module parameters
cons_listen 0.0.0.0:5555
http_listen 0.0.0.0:8000
evdev_device /dev/input/event0
# Speex codec parameters
speex_quality 7 # 0-10
speex_complexity 7 # 0-10
speex_enhancement 0 # 0-1
speex_mode_nb 3 # 1-6
speex_mode_wb 6 # 1-6
speex_vbr 0 # Variable Bit Rate 0-1
speex_vad 0 # Voice Activity Detection 0-1
speex_agc_level 8000
# Opus codec parameters
opus_bitrate 28000 # 6000-510000
# NAT Behavior Discovery
natbd_server creytiv.com
natbd_interval 600 # in seconds
# Selfview
video_selfview window # {window,pip}
#selfview_size 64x64
# ICE
ice_turn no
ice_debug no
ice_nomination regular # {regular,aggressive}
ice_mode full # {full,lite}
from baresip.
You can also test with audio-loop (command "a"). Do you use a clean alsa or with pulseaudio?
from baresip.
I tried the audio-loop command as you suggested, but I have no sound. It seems that pulseaudio is used when I use baresip; I don't know much regarding audio configuration, but based on this answer on Ask Ubuntu, I did grep the ouput of pactl list
:
$ pactl list | grep baresip
application.name = "ALSA plug-in [baresip]"
application.process.binary = "baresip"
module-stream-restore.id = "sink-input-by-application-name:ALSA plug-in [baresip]"
application.name = "ALSA plug-in [baresip]"
application.process.binary = "baresip"
module-stream-restore.id = "source-output-by-application-name:ALSA plug-in [baresip]"
application.name = "ALSA plug-in [baresip]"
application.process.binary = "baresip"
application.name = "ALSA plug-in [baresip]"
application.process.binary = "baresip"
I can try without PulseAudio but I have to find out how to disable it for Baresip first... The output of baresip when invoking audio-loop gives:
Audio-loop: 8000Hz, 1ch
8000Hz 1ch n_read=238 n_write=244 rw_ratio=0.97audio-loop stopped
However, joining the thread does not block when quitting the application!
from baresip.
I just tried audio-loop when running baresip without pulseaudio (using pasuspender
as mentioned here). I have no sound, and the application hangs when exiting; the backtrace is slightly different but leads to the same code:
(gdb) bt
#0 0x00007ffff6e664db in pthread_join (threadid=140737122633472, thread_return=0x0) at pthread_join.c:92
#1 0x00007ffff53504b6 in ausrc_destructor (arg=0x671790) at modules/alsa/alsa_src.c:40
#2 0x00007ffff7bbd584 in mem_deref (data=0x671790) at src/mem/mem.c:318
#3 0x00007ffff4441e21 in auloop_destructor (arg=0x660430) at modules/auloop/auloop.c:75
#4 0x00007ffff7bbd584 in mem_deref (data=0x660430) at src/mem/mem.c:318
#5 0x00007ffff4442851 in auloop_stop (pf=0x7ffff5b57550 <pf_stderr>, arg=0x7fffffffdd90) at modules/auloop/auloop.c:349
#6 0x00000000004113c2 in cmd_process (ctxp=0x6299f0 <uictx>, key=65 'A', pf=0x7ffff5b57550 <pf_stderr>) at src/cmd.c:302
#7 0x000000000041bd43 in ui_handler (key=65 'A', pf=0x7ffff5b57550 <pf_stderr>) at src/ui.c:17
#8 0x000000000041be40 in ui_input_key (key=65 'A', pf=0x7ffff5b57550 <pf_stderr>) at src/ui.c:83
#9 0x00007ffff5956d0f in report_key (ui=0x655ae0, key=65 'A') at modules/stdio/stdio.c:66
#10 0x00007ffff5956d9f in ui_fd_handler (flags=1, arg=0x655ae0) at modules/stdio/stdio.c:90
#11 0x00007ffff7bbb53f in fd_handler (re=0x7ffff7ddbcc0 <global_re>, fd=0, flags=1) at src/main/main.c:246
#12 0x00007ffff7bbc4f1 in fd_poll (re=0x7ffff7ddbcc0 <global_re>) at src/main/main.c:849
#13 0x00007ffff7bbc7ee in re_main (signalh=0x41f29b <signal_handler>) at src/main/main.c:992
#14 0x000000000041f54b in main (argc=1, argv=0x7fffffffe1b8) at src/main.c:169
from baresip.
With pulseaudio 6.0 on archlinux there is no such problem. You can check with "pavucontrol" which audio interface is mapped to baresip. Maybe I can check this with debian jessie and pulseaudio next week too.
from baresip.
most likely the read thread is hanging here:
while (st->run) {
err = snd_pcm_readi(st->read, st->sampv, num_frames);
in blocking mode this function call will not return until there are samples in the buffer.
so it you have problems with your Audio device then we might have to wait forever :)
could you try a couple of things:
- use NONBLOCK mode
in alsa_src.c:
- err = snd_pcm_open(&st->read, st->device, SND_PCM_STREAM_CAPTURE, 0);
+ err = snd_pcm_open(&st->read, st->device, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
- close the audio-device BEFORE calling pthread_join, in destructor:
static void ausrc_destructor(void *arg)
{
struct ausrc_st *st = arg;
if (st->read)
snd_pcm_close(st->read);
/* Wait for termination of other thread */
if (st->run) {
st->run = false;
(void)pthread_join(st->thread, NULL);
}
mem_deref(st->sampv);
mem_deref(st->device);
}
I have seen similar bugs before, so most likely it is a ALSA API-usage error (i.e. baresip).
NOTE: using non-blocking mode is not really an option as it consumes 100% cpu
from baresip.
from baresip.
is there any more updates on this one?
if not feel free to close the Issue.
from baresip.
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from baresip.