May I ask how to set the sampling rate of saved audio files to 16000, with a default of 8000

[aonoa]

G711 encoding used

stream: update 'audio'
audio: Set audio decoder: PCMU 8000Hz 1ch
audio: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch

[aonoa]

There seems to be a problem here

[aonoa]

module			auresamp.so
auplay_srate            16000

I set it this way, but the audio is distorted and the playback speed is twice as fast as before

[aonoa]

I'm not sure if this is because I configured it incorrectly or if it's a bug

[cspiel1]

I'll have a look on this soon.

[cspiel1]

Did you use module aufile to write received audio stream to a WAV file? Please attach your .baresip/config!

So long I'll :

• try aufile with auresamp 16000 Hz
• check/correct the Wiki

[aonoa]

#
# baresip configuration
#

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

# SIP
#sip_listen		0.0.0.0:5060
#sip_certificate	cert.pem
sip_cafile		/etc/ssl/certs/ca-certificates.crt
#sip_transports		udp,tcp,tls,ws,wss
#sip_trans_def		udp
#sip_verify_server	yes
sip_tos			160

# Call
call_local_timeout	120
call_max_calls		4
call_hold_other_calls	yes

# Audio
#audio_path		/usr/local/share/baresip
audio_player		alsa,default
audio_source		alsa,default
audio_alert		alsa,default
#ausrc_srate		48000
#auplay_srate		48000
auplay_srate		16000
#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
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/video0
#video_display		x11,nil
video_size		640x480
video_bitrate		1000000
video_fps		30.00
video_fullscreen	no
videnc_format		yuv420p

# 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_delay	5-10		# (min. frames)-(max. packets)
video_jitter_buffer_type	fixed		# off, fixed, adaptive
video_jitter_buffer_delay	5-50		# (min. frames)-(max. packets)
rtp_stats		no
#rtp_timeout		60
#avt_bundle		no
#rtp_rxmode		main

# Network
#dns_server		1.1.1.1:53
#dns_server		1.0.0.1:53
#dns_fallback		8.8.8.8:53
#dns_getaddrinfo		no
#net_interface		enp0s3

# Play tones
#file_ausrc		aufile
#file_srate		16000
#file_channels		1

#------------------------------------------------------------------------------
# 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			amr.so
#module			g7221.so
#module			g722.so
#module			g726.so
module			g711.so
#module			l16.so
#module			mpa.so
#module			codec2.so

# Audio filter Modules (in encoding order)
module			auconv.so
module			auresamp.so
#module			vumeter.so
module			sndfile.so
module			pcmfile.so
#module			plc.so
#module			webrtc_aec.so

# Audio driver Modules
module			alsa.so
#module			pulse.so
#module			pipewire.so
#module			jack.so
#module			portaudio.so
#module			aubridge.so
#module			aufile.so
#module			ausine.so

# Video codec Modules (in order)
#module			avcodec.so
#module			vp8.so
#module			vp9.so

# Video filter Modules (in encoding order)
#module			selfview.so
#module			snapshot.so
#module			swscale.so
#module			vidinfo.so
#module			avfilter.so

# Video source modules
#module			v4l2.so
#module			vidbridge.so

# Video display modules
#module			directfb.so
#module			x11.so
#module			sdl.so
#module			fakevideo.so

# Audio/Video source modules
#module			avformat.so
#module			gst.so

# Compatibility modules
#module			ebuacip.so
module			uuid.so

# Media NAT modules
module			stun.so
module			turn.so
module			ice.so
#module			natpmp.so
#module			pcp.so

# Media encryption modules
#module			srtp.so
#module			dtls_srtp.so
#module			gzrtp.so


#------------------------------------------------------------------------------
# Application Modules

module_app		account.so
module_app		contact.so
module_app		debug_cmd.so
#module_app		echo.so
#module_app		gtk.so
module_app		menu.so
#module_app		mwi.so
#module_app		presence.so
#module_app		serreg.so
#module_app		syslog.so
#module_app		mqtt.so
#module_app		ctrl_tcp.so
#module_app		ctrl_dbus.so
#module_app		httpreq.so
#module_app		multicast.so
module_app		netroam.so


#------------------------------------------------------------------------------
# Module parameters

# DTLS SRTP parameters
#dtls_srtp_use_ec	prime256v1


# UI Modules parameters
cons_listen		0.0.0.0:5555 # cons - Console UI UDP/TCP sockets

http_listen		0.0.0.0:8000 # httpd - HTTP Server

ctrl_tcp_listen		0.0.0.0:4444 # ctrl_tcp - TCP interface JSON

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 (expected packet loss)

# 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

# Menu
#redial_attempts	0 # Num or <inf>
#redial_delay		5 # Delay in seconds
#ringback_disabled	no
#statmode_default	off
#menu_clean_number	no
#sip_autoanswer_method	rfc5373 # {rfc5373,call-info,alert-info}
#ring_aufile		ring.wav
#callwaiting_aufile	callwaiting.wav
#ringback_aufile	ringback.wav
#notfound_aufile	notfound.wav
#busy_aufile		busy.wav
#error_aufile		error.wav
#sip_autoanswer_aufile	autoanswer.wav
#menu_max_earlyaudio	32
#menu_max_earlyvideo_rx	32
#menu_max_earlyvideo_tx	32

# GTK
#gtk_clean_number	no

# avcodec
#avcodec_h264enc	libx264
#avcodec_h264dec	h264
#avcodec_h265enc	libx265
#avcodec_h265dec	hevc
#avcodec_hwaccel	vaapi
#avcodec_profile_level_id 42002a
#avcodec_keyint		10

# ctrl_dbus
#ctrl_dbus_use	system		# system, session

# mqtt
#mqtt_broker_host	sollentuna.example.com
#mqtt_broker_port	1883
#mqtt_broker_cafile	/path/to/broker-ca.crt	# set this to enforce TLS
#mqtt_broker_clientid	baresip01	# has to be unique
#mqtt_broker_user	user
#mqtt_broker_password	pass
#mqtt_basetopic		baresip/01

# sndfile
snd_path		/tmp
pcm_path		/tmp

# EBU ACIP
#ebuacip_jb_type	fixed	# auto,fixed

# HTTP request module
#httpreq_ca		trusted1.pem
#httpreq_ca		trusted2.pem
#httpreq_dns		1.1.1.1
#httpreq_dns		8.8.8.8
#httpreq_hostname	myserver
#httpreq_cert		cert.pem
#httpreq_key		key.pem

# multicast receivers (in priority order)- port number must be even
#multicast_call_prio	0
#multicast_ttl	1
#multicast_jbuf_type	fixed		# off, fixed, adaptive
#multicast_jbuf_delay	5-10		# frames
#multicast_listener	224.0.2.21:50000
#multicast_listener	224.0.2.21:50002

# avformat
#avformat_hwaccel	vaapi
#avformat_inputformat	mjpeg
#avformat_decoder	mjpeg
#avformat_pass_through	yes
#avformat_rtsp_transport	udp

# ice
#ice_policy		all	# all, relay (candidates)

[cspiel1]

Updated the Wiki. It was wrong.

auplay_srate sets the sample rate of the audio player.
Additionally it is used by some audio filters (mostly for the output audio format). I am not sure if it is used everywhere correctly. I'll plan to verify at least mixausrc.

1. Writing WAV files can be done by means of the audio filter module sndfile. I'll test/verify this with different settings for the sample rate and with different codecs (G.711, G.722, opus).

2. With module aufile the received audio stream can be written to a WAV file instead of playing to the speaker. If auresamp is loaded every combination of codec (sample rate) and auplay_srate should be possible. I'll test this.

[cspiel1]

Tests

Write speaker.wav with aufile

config:

audio_player       aufile,
module		   aufile.so
module             auresamp.so

module             opus.so
module             g722.so
module             g711.so

• opus with sample rate 48000 -> writes speaker.wav PCM 16bit mono with 48000 Hz -> Ok
• G722 with 16000 -> writes speaker.wav mono 16000 Hz -> Ok
• G711 with 8000 -> writes speaker.wav mono 8000 Hz -> Ok
• opus with 48000, auplay_srate 32000-> "auresamp_setup error" -> Not supported (The resampler does not support this)
• opus with 48000, auplay_srate 24000-> writes speaker.wav with mono 24000 Hz -> Ok
• G722 with 16000, auplay_srate 32000-> Ok (upsampling also works)

Use sndfile filter

module      resample.so
module      sndfile.so

• opus, auplay_srate not set -> writes dump-sip...wav file with 48000 Hz -> Ok
• G722, auplay_srate not set -> writes dump-sip...wav file with 16000 Hz -> Ok

But ffplay can't play it on my machine. paplay plays these files. In audacity they look okay. Have to look closer.

[aonoa]

Thanks ，I use sndfile filter save

[cspiel1]

Use sndfile filter - proceeding

• G711 with 8000, auplay_srate 16000 -> writes a WAV file with 16000Hz header but it seems to have only 8000Hz -> NOK

Thus indeed there seems to be an issue.

[cspiel1]

If one want to get a correct resampled WAV file, then the solution is to change the order of the filters.

Wrong:

module			auresamp.so
module			auconv.so
module			sndfile.so

Correct:

module			sndfile.so
module			auresamp.so
module			auconv.so

Reason:
The filter module sndfile does not support resampling on its own. It depends on auresamp for resampling and on auconv for sample format conversion.
And the filters for the decoded (incoming) audio stream are processed in reverse order. Thus correct is: First do the conversion and resampling, then write the WAV file.

Remaining issue:
If the wrong order is configured, then sndfile should write the correct sample rate into the WAV header. I'll try to find a bugfix.

Bugfix: #2976

[cspiel1]

ffplay

Doesn't like the ':' and maybe some other symbols in the file name. But it can be played by specifying the absolute path and putting the path into single quotes.

ffplay  '/home/cspiel/src/baresip/baresip/dump-sip:A@localhost=>sip:10.1.0.199-2024-03-20-09-02-34-dec.wav'

[cspiel1]

Also found sample rate issues in mixausrc.

Bugfix: #2981