audio problems in Windows

[jobo-zt]

The master code is tested in Windows 10, with the following problems.（test winows10 baresip <-> linux(RPI) baresip）

1. The local audio has no code to spread to the remote.
2. During the meeting, the local cannot hang up normally. (neither remote nor local)

//log
PS D:\baresip\bin_vpn> .\baresip-win32.exe -e "/dial sip:0755000002@10.8.0.2"
baresip v0.6.5 Copyright (C) 2010 - 2020 Alfred E. Heggestad et al.
Binding to interface or IP address '10.8.0.3'
Local network address: IPv4={E6BEA673-6CFE-46B8-A7E8-B6CEB6F98FBB}|10.8.0.3
httpd: listening on 0.0.0.0:8000
aucodec: opus/48000/2
aucodec: PCMU/8000/1
aucodec: PCMA/8000/1
aufilt: speex_pp
winwave: output devices: 1, input devices: 1
ausrc: winwave
auplay: winwave
vidcodec: H264
vidcodec: H264
vidcodec: H263
vidcodec: MP4V-ES
vidcodec: H265
avcodec: using H.264 encoder 'libx264' -- libx264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
avcodec: using H.264 decoder 'h264' -- H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
avcodec: using H.265 encoder 'libx265' -- libx265 H.265 / HEVC
avcodec: using H.265 decoder 'hevc' -- HEVC (High Efficiency Video Coding)
vidsrc: dshow
vidisp: sdl
medianat: stun
medianat: turn
medianat: ice
medianat: ice-lite
Populated 1 account
glog: module init
Could not create logging file: No such file or directory
Could not create logging file: No such file or directory
Populated 3 contacts
Populated 3 audio codecs
Populated 1 audio filter
Populated 5 video codecs
Populated 0 video filters
baresip is ready.
/dial sip:0755000002@10.8.0.2
ua: using best effort AF: af=AF_INET
call: connecting to 'sip:0755000002@10.8.0.2'..
unhandeled request from 10.8.0.3:5060: REGISTER sip:10.8.0.3
reg: sip:0755000003@10.8.0.3: 501 Not Implemented (baresip v0.6.5 (?/win32))
call: SIP Progress: 180 Ringing (/)
play: /usr/share/baresip/ringback.wav: No such file or directory
stream: incoming rtp for 'audio' established, receiving from 10.8.0.2:18348
audio: Set audio decoder: opus 48000Hz 2ch
audio: player started with sample format S16LE
long async blocking: 240>100 ms (h=0xfc9f00 arg=0x1592e38)
stream: update 'audio'
stream: update 'video'
audio: Set audio encoder: opus 48000Hz 2ch
speex_pp: Speex preprocessor loaded: srate = 48000Hz
audio: source started with sample format S16LE
audio tx pipeline: winwave ---> speex_pp ---> opus
audio rx pipeline: winwave <--- opus
Set video encoder: H264 packetization-mode=0 (2000000 bit/s, 25.00 fps)
avcodec: h264 encoder activated
dshow: got device '' id=0
Set video decoder: H264 packetization-mode=0
avcodec: h264 decoder activated
avcodec: decode: hardware accel disabled
0755000003@10.8.0.3: Call established: sip:0755000002@10.8.0.2
long async blocking: 715>100 ms (h=0xfc9f00 arg=0x1593180)
stream: incoming rtp for 'video' established, receiving from 10.8.0.2:48144
[h264 @ 038e8bc0] no frame!
avcodec: avcodec_send_packet error, packet=26 bytes, ret=-1094995529 (Invalid data found when processing input)
video: H264 decode error (seq=9012, 5 bytes): Unknown error
video: receiving with resolution 352 x 288 and format 'yuv420p'
sdl: created renderer 'direct3d'
long async blocking: 308>100 ms (h=0xfc9f00 arg=0x15a0890)
[libx264 @ 038d6940] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 038d6940] profile Constrained Baseline, level 3.1, 4:2:0, 8-bit
long async blocking: 762>100 ms (h=0xf81760 arg=0x158fcd8).8/22.0
long async blocking: 102>100 ms (h=0xfad4c0 arg=0x15e1ec0).7/22.8
unhandeled request from 10.8.0.3:5060: REGISTER sip:10.8.0.34/21.0
reg: sip:0755000003@10.8.0.3: 501 Not Implemented (baresip v0.6.5 (?/win32))
long async blocking: 108>100 ms (h=0xfad4c0 arg=0x15e1ec0)0.5/22.8
long async blocking: 617>100 ms (h=0xf7b950 arg=(nil))ps=10.0/19.0
long async blocking: 108>100 ms (h=0xf7b950 arg=(nil))ps=10.0/19.0
[0:01:43] audio=0/28390 video=506620/281671 (bit/s) efps=10.0/14.5

[alfredh]

hi @75535552

I just wanted to check if you are a real person and not a robot.

Are you working for a company that is using the product ?

if the request for help is coming from a real person, it is easier for me
to help you.

if the request is coming from 981723981264912763912837 I am not really sure what to do :)

Alfred

[jobo-zt]

hi @alfredh
I'm a multimedia communications developer and I'm sure it's not a robot. It's not used in my project. Baresip implementation is very good. I like it in my spare time. I put it in RPI, which can connect with the Windows client very well. when the code is updated, I will test it in windows and Linux. about "981723981264912763912837" I don't quite understand what line and column marks my from. I don't think it's valid

[alfredh]

if you have signed your request with a real name, it would become more personal,
and hence easier for me to help you.

if I receive a bug report from 12345 it is not really easy for me to take it seriously.
it can be a automated bot.

regarding your issue:

• what OS version are you running (exact) ?

• what kind of audio hardware do you use ?

• please share the full config+accounts

• have you tried if the /auloop command is working?

• please dump the output of /audio_debug command after 10 second in the call

• add some debug code here and check if samples are coming:

https://github.com/alfredh/baresip/blob/master/modules/winwave/src.c#L81

• this line: "glog: module init" is not from stock baresip. Please use a clean baresip
  from git HEAD without any local modifications when reporting.

[jobo-zt]

Thanks. I'll update it later.

[jobo-zt]

hi @alfredh

I git reset --hard *** without any. in Win10 enterprise x64 IDE is vs2015 thinkpad t470s

I went back to the code to confirm that the problem was introduced in the following submission. The previous version is OK.
//
Revision: 3f2a182
Author: Ola Palm ola.palm@me.com
Date: 2020/4/4 15:43:41
Message:
Aubuf current jb value to mqtt (#954)

• Aubuf current jb value to mqtt
• fix comments.
• changed type for current_jb value to uint32
• another type update to get by clang on mac
• tabs not spaces

---

Modified: include/baresip.h
Modified: modules/mqtt/publish.c
Modified: src/audio.c
Modified: src/event.c

//config

baresip configuration

#------------------------------------------------------------------------------

Core

poll_method select # poll, select

SIP

sip_listen 0.0.0.0:5060
#sip_certificate cert.pem
#sip_cafile /etc/ssl/certs/ca-certificates.crt

Call

call_local_timeout 120
call_max_calls 4

Audio

#audio_path /usr/share/baresip
audio_player winwave,nil
audio_source winwave,nil
audio_alert winwave,nil
#ausrc_srate 48000
#auplay_srate 48000
#ausrc_channels 0
#auplay_channels 0
#audio_txmode poll # poll, thread
audio_level no
ausrc_format s16 # s16, float, ..
auplay_format s16 # s16, float, ..
auenc_format s16 # s16, float, ..
audec_format s16 # s16, float, ..
audio_buffer 20-160 # ms

Video

#video_source dshow,nil
#video_display sdl,nil
video_size 352x288
video_bitrate 500000
video_fps 25.00
video_fullscreen no
videnc_format yuv420p

AVT - Audio/Video Transport

rtp_tos 184
#rtp_ports 10000-20000
#rtp_bandwidth 512-1024 # [kbit/s]
rtcp_mux no
jitter_buffer_delay 5-10 # frames
rtp_stats no
#rtp_timeout 60

Network

#dns_server 1.1.1.1:53
#dns_server 1.0.0.1:53
#net_interface eth0
#net_interface 192.168.9.100
net_interface 10.8.0.3

#BFCP
bfcp_proto udp

#------------------------------------------------------------------------------

Modules

#module_path /usr/lib/baresip/modules

UI Modules

module wincons.dll
#module cons.dll
#module evdev.dll
#module httpd.dll

Audio codec Modules (in order)

#module opus.dll
#module amr.dll
#module g7221.dll
#module g722.dll
#module g726.dll
module g711.dll
#module gsm.dll
#module l16.dll
#module mpa.dll
#module codec2.dll
#module ilbc.dll
#module isac.dll

Audio filter Modules (in encoding order)

#module vumeter.dll
#module sndfile.dll
#module speex_pp.dll
#module plc.dll
#module webrtc_aec.dll

Audio driver Modules

module winwave.dll
#module jack.dll
#module portaudio.dll
#module aubridge.dll
#module aufile.dll

Video codec Modules (in order)

#module avcodec.dll
#module vp8.dll
#module vp9.dll

Video filter Modules (in encoding order)

#module selfview.dll
#module snapshot.dll
#module swscale.dll
#module vidinfo.dll

Video source modules

module dshow.dll
#module x11grab.dll
#module cairo.dll
#module vidbridge.dll

Video display modules

#module x11.dll
#module sdl.dll
#module fakevideo.dll

Audio/Video source modules

#module avformat.dll
#module rst.dll
#module gst.dll
#module gst_video.dll

Compatibility modules

#module ebuacip.dll

Media NAT modules

module stun.dll
module turn.dll
module ice.dll
#module natpmp.dll
#module pcp.dll

Media encryption modules

#module srtp.dll
#module dtls_srtp.dll
#module zrtp.dll

#------------------------------------------------------------------------------

Temporary Modules (loaded then unloaded)

module_tmp uuid.dll
module_tmp account.dll

#------------------------------------------------------------------------------

Application Modules

module_app auloop.dll
#module_app b2bua.dll
module_app contact.dll
module_app debug_cmd.dll
#module_app echo.dll
#module_app gtk.dll
module_app menu.dll
#module_app mwi.dll
#module_app presence.dll
#module_app syslog.dll
#module_app mqtt.dll
#module_app ctrl_tcp.dll
module_app vidloop.dll

#------------------------------------------------------------------------------

Module parameters

UI Modules parameters

cons_listen 0.0.0.0:5555 # cons

http_listen 0.0.0.0:8000 # httpd - server

ctrl_tcp_listen 0.0.0.0:4444 # ctrl_tcp

evdev_device /dev/input/event0

Opus codec parameters

opus_bitrate 28000 # 6000-510000
#opus_stereo yes
#opus_sprop_stereo yes
#opus_cbr no
#opus_inbandfec no
#opus_dtx no
#opus_mirror no
#opus_complexity 10
#opus_application audio # {voip,audio}
#opus_samplerate 48000
#opus_packet_loss 10 # 0-100 percent

Opus Multistream codec parameters

#opus_ms_channels 2 #total channels (2 or 4)
#opus_ms_streams 2 #number of streams
#opus_ms_c_streams 2 #number of coupled streams

vumeter_stderr yes

#jack_connect_ports yes

Selfview

video_selfview window # {window,pip}
#selfview_size 64x64

ICE

ice_debug no

ZRTP

#zrtp_hash no # Disable SDP zrtp-hash (not recommended)

Menu

#menu_bell yes
#redial_attempts 0 # Num or
#redial_delay 5 # Delay in seconds
#ringback_disabled no
#statmode_default off

avcodec

#avcodec_h264enc libx264
#avcodec_h264dec h264
#avcodec_h265enc libx265
#avcodec_h265dec hevc
#avcodec_hwaccel none

mqtt

#mqtt_broker_host 127.0.0.1
#mqtt_broker_port 1883
#mqtt_broker_clientid baresip01
#mqtt_broker_user user
#mqtt_broker_password pass
#mqtt_basetopic baresip/01

sndfile

#snd_path /tmp

EBU ACIP

#ebuacip_jb_type fixed # auto,fixed

//accounts
sip:0755000003@10.8.0.3;auth_pass=0755000003

I will continue to provide other info.

[jobo-zt]

hi @alfredh @ola Palm
I found waveInGetPosition function in waveInCallback.

[waveInGetPosition is no longer supported for use as of Windows Vista. Instead, use IAudioClock::GetPosition
https://docs.microsoft.com/zh-cn/windows/win32/api/mmeapi/nf-mmeapi-waveingetposition?redirectedfrom=MSDN

[alfredh]

I pushed a fixed that uses tmr_jiffies_usec instead

if you want to use IAudioClock::GetPosition then please make a PR for that.

Many thanks for reporting the bug!