Question about audio loop

Vladimir B

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Dec 4, 2025, 6:56:23 AM (8 days ago) Dec 4

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Hello, iam start using Baresip recently. I have a question about audio loop. I used baresip like a client to connect to my audio server, all works fine and i can send some wav file and check it on server. But i need to loop this wav file for its played in a circle. How i can do that? I heard that in previos versions it was possible with settings like: avformat_stream_loop 1

so i need some help with that if its possible.....

my config:

baresip configuration
#

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

# SIP
#sip_listen 0.0.0.0:5060
#sip_certificate cert.pem
sip_cafile /etc/ssl/certs/ca-certificates.crt
sip_capath /etc/ssl/certs
#sip_transports udp,tcp,tls,ws,wss
#sip_trans_def udp
#sip_verify_server yes
#sip_verify_client no
#sip_tls_resumption all
sip_tos 160
#filter_registrar udp,tcp,tls,ws,wss

# Call
call_local_timeout 120
call_max_calls 4
call_hold_other_calls yes
call_accept no

# Audio
#audio_path /usr/local/share/baresip
#audio_player alsa,default
audio_source aufile,/home/borovskii/Downloads/sample.wav
#audio_source aufile,/home/borovskii/Downloads/poisonruin.mp3
#audio_alert alsa,default
ausrc_srate 48000
#auplay_srate 48000
ausrc_channels 2
#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
#audio_buffer_mode fixed # fixed, adaptive
#audio_silence -35.0 # in [dB]
#audio_telev_pt 101 # payload type for telephone-event

# Video
#video_source v4l2,/dev/video
#video_source h.264,/home/borovskii/Documents/baresip/IAmLegendTrailerFullHD.h264
#video_display x11,nil
#video_size 1920x1080
#video_bitrate 4009600
#video_fps 30.00
#video_fullscreen yes
#videnc_format yuv420p
#videnc_format libx264

# AVT - Audio/Video Transport
rtp_tos 184
rtp_video_tos 136
#rtp_ports 10000-20000
#rtp_bandwidth 512-1024 # [kbit/s]
audio_jitter_buffer_type fixed # off, fixed, adaptive
audio_jitter_buffer_ms 100-200 # Min. - Max. [ms]
audio_jitter_buffer_size 50 # [packets]
video_jitter_buffer_type fixed # off, fixed, adaptive
video_jitter_buffer_ms 100-200 # Min. - Max. [ms]
video_jitter_buffer_size 250 # [packets]
rtp_stats no
#rtp_timeout 60
#avt_bundle no
#rtp_rxmode main

Vladimir B

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Dec 4, 2025, 7:22:08 AM (8 days ago) Dec 4

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and if i remember correct it was libraris nemed auloop.so and vidloop.so in previous versions.......

четверг, 4 декабря 2025 г. в 14:56:23 UTC+3, Vladimir B:

Alfred E. Heggestad

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Dec 4, 2025, 8:31:42 AM (8 days ago) Dec 4

to Vladimir B, baresip

Hi

Please try this module:

https://github.com/baresip/baresip-apps/tree/main/modules/auloop

Alfred

On 04/12/2025 13:22, Vladimir B wrote:
> and if i remember correct it was libraris nemed auloop.so and vidloop.so
> in previous versions.......
> четверг, 4 декабря 2025 г. в 14:56:23 UTC+3, Vladimir B:
>
> Hello, iam start using Baresip recently. I have a question about
> audio loop. I used baresip like a client to connect to my audio
> server, all works fine and i can send some wav file and check it on
> server. But i need to loop this wav file for its played in a circle.
> How i can do that? I heard that in previos versions it was possible
> with settings like: avformat_stream_loop 1
> so i need some help with that if its possible.....
>
> my config:
>
> baresip configuration
> #
>
> #------------------------------------------------------------------------------
>
> # SIP

> #sip_listen 0.0.0.0:5060 <http://0.0.0.0:5060>

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> a383f5499962n%40googlegroups.com?utm_medium=email&utm_source=footer>.

Vladimir B

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Dec 4, 2025, 8:41:53 AM (8 days ago) Dec 4

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Can u plz tell me more details). If i understand correct first off all i need to install this module? And how i can do that?

четверг, 4 декабря 2025 г. в 16:31:42 UTC+3, alfred.h...@gmail.com:

Vladimir B

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Dec 4, 2025, 9:02:09 AM (8 days ago) Dec 4

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Iam download https://github.com/baresip/baresip-apps unpack it and just make:

$ cd baresip $ cmake -B build $ cmake --build build $ cmake --install build

now i have in my /usr/local/baresip/modules/auloop.so, vidloop.so

am i correct?

четверг, 4 декабря 2025 г. в 16:41:53 UTC+3, Vladimir B:

Vladimir B

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Dec 4, 2025, 9:45:49 AM (8 days ago) Dec 4

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ok, its look like auloop is added

ESC Hangup call
/100rel .. Set 100rel mode
/about About box
/accept a Accept incoming call
/acceptdir .. Accept incoming call with audio and videodirection.
/addcontact .. Add a contact
/answermode .. Set answer mode
/apistate User Agent state
/aufileinfo .. Audio file info
/auloop .. Start audio-loop <srate ch>
/auloop_stop Stop audio-loop
/auplay .. Switch audio player
/ausrc .. Switch audio source
/callstat c Call status
/conf_reload Reload config file
/config Print configuration
/contact_next > Set next contact
/contact_prev < Set previous contact
/contacts C List contacts
/dial .. d .. Dial
/dialcontact D Dial current contact
/dialdir .. Dial with audio and videodirection.
/dnd .. Set Do not Disturb
/entransp .. Enable/Disable transport
/hangup b Hangup call
/hangupall .. Hangup all calls with direction
/help h Help menu
/insmod .. Load module
/listcalls l List active calls
/loglevel v Log level toggle
/main Main loop debug
/memstat y Memory status
/message .. M .. Message current contact
/modules Module debug
/netchange Inform netroam about a network change
/netstat n Network debug
/options .. o .. Options
/play .. Play audio file
/quit q Quit
/refer .. R .. Send REFER outside dialog
/reginfo r Registration info
/rmcontact .. Remove a contact
/rmmod .. Unload module
/setadelay .. Set answer delay for outgoing call
/setansval .. Set value for Call-Info/Alert-Info
/sipstat i SIP debug
/siptrace .. SIP trace
/sysinfo s System info
/timers Timer debug
/tlsissuer TLS certificate issuer
/tlssubject TLS certificate subject
/uaaddheader .. Add custom header to UA
/uadel .. Delete User-Agent
/uadelall .. Delete all User-Agents
/uafind .. Find User-Agent <aor>
/uanew .. Create User-Agent
/uareg .. UA register <regint> [index]
/uarmheader .. Remove custom header from UA
/uastat u UA debug
/uuid Print UUID
/vidsrc .. Switch video source

/auloop
Usage: /auloop <samplerate> <channels>

But how i can use it? what means samplerate and channels?

четверг, 4 декабря 2025 г. в 17:02:09 UTC+3, Vladimir B:

Vladimir B

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Dec 4, 2025, 10:07:32 AM (8 days ago) Dec 4

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Ok i used rate and channels but for some reason the loop is stop with this error:

/auloop <48000 2>
Audio-loop: 48000Hz, 2ch, S16LE
alsa: reset: srate=48000, ch=2, num_frames=960, pcmfmt=S16_LE
alsa: playback started (default)
aufile: loading input file '/home/borovskii/Downloads/sample.wav'
aufile: /home/borovskii/Downloads/sample.wav: 8000 Hz, 1 channels, S16LE
aufile: audio ptime=20 sampc=160
aufile: read end of file
aufile: loaded 150400 bytes
aufile: end of filen_read=0.776 n_write=9.400 rw_delay=-8.623 [sec] rw_ratio=0.082624
auloop: ausrc error: Success [0] (end of file)
~~~~~ Audioloop summary: ~~~~~
48000 Hz 2ch S16LE

* Source
module aufile
samples 75200
duration 0.783 sec
frames 470 (avg ptime 1.66ms)

* Aubuf
overrun 0
underrun 437

alsa: stopping playback thread (default)
* Player
module alsa
samples 908160
duration 9.460 sec
frames 473 (avg ptime 20.00ms)

четверг, 4 декабря 2025 г. в 17:45:49 UTC+3, Vladimir B:

Vladimir B

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Dec 4, 2025, 10:56:23 AM (8 days ago) Dec 4

to baresip

but if there is no any files to loop it works ok...

/auloop <48000 2>
Audio-loop: 48000Hz, 2ch, S16LE
alsa: reset: srate=48000, ch=2, num_frames=960, pcmfmt=S16_LE
alsa: playback started (default)
alsa: reset: srate=48000, ch=2, num_frames=960, pcmfmt=S16_LE

alsa: recording started (default) format=S16LE
48000Hz 2ch S16LE n_read=114.159 n_write=114.299 rw_delay=-0.140 [sec] rw_ratio=0.998775

четверг, 4 декабря 2025 г. в 18:07:32 UTC+3, Vladimir B:

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