I will try to use the loopback module since it performs adaptive resampling between source and sink. If I understand what you write, I should use something like this : $ pacmd load-module module-null-sink sink_name=MySink $ pacmd load-module module-null-sink sink_name=MySink2 $ pacmd load-module module-loopback source=alsa_input.usb-0d8c_C- Media_USB_Headphone_Set-00.mono-fallback sink=MySink2 And then start my two gstreamer pipelines : $ gst-launch-1.0 pulsesrc device=MySink2.monitor ! audioconvert ! audioresample ! pulsesink device=MySink $ gst-launch-1.0 pulsesrc device=MySink.monitor ! audio/x- raw,channels=2 ! audioconvert ! audioresample ! opusenc bitrate=256000 ! oggmux ! shout2send ip=... port=... mount=... password=... Is it correct ? At this moment, I don't try to mesure latency between audio input and output. ++ Jack Le dimanche 13 février 2022 à 20:17 +0100, Georg Chini a écrit : > On 13.02.22 17:09, corbeille wrote: > > Hey Georg, > > > > I have updated my to raspios since last time : Debian GNU/Linux 11 > > (bullseye). > > It comes with : > > - pulseaudio 14.2 > > - GStreamer 1.18.4 > > > > Is it also too old ? > > I was hoping to avoid installing pulseaudio from source. > > If so, I will give it a try... > > ++ > > > > Jack > > Well, 14.2 is at least not completely out of date. Nevertheless I > would try > current git. What I wonder about is that the problem only occurs with > a > second null sink. Are you sure about that? The two null sinks should > not > interact at all. Maybe with the second null sink the problem occurs > only > later? > > I can understand that there is an issue, because the system time and > sound > card time are normally different. So if the source for example (from > a > system > time perspective) delivers samples at 48003 Hz while the samples are > played > with 48000 Hz you will have three samples left per second which pile > up > pretty fast. module-loopback has mechanisms to deal with the rate > difference > between source and sink. Maybe you can try to use a loopback from the > alsa > source to a null-sink and then use the monitor of that null sink in > your > first > gst-launch command instead of using the alsa source directly. > > Do you see increasing latency before the sound gets choppy? > > > > > Le dimanche 13 février 2022 à 17:05 +0100, Georg Chini a écrit : > > > On 13.02.22 16:57, corbeille wrote: > > > > Hello, > > > > > > > > I did some additional tests. > > > > And it seems the problem occurs when I have a second null sink. > > > > With > > > > only one null sink, I don't have any problem. Weird. > > > > > > > > So here is an example of the configuration that causes the > > > > problem > > > > : > > > > > > > > 1st shell: > > > > $ pacmd load-module module-null-sink sink_name=MySink > > > > $ pacmd load-module module-null-sink sink_name=MySink2 > > > > > > > > 2nd shell: > > > > $ gst-launch-1.0 pulsesrc device="alsa_input.usb-0d8c_C- > > > > Media_USB_Headphone_Set-00.analog-mono" ! audioconvert ! > > > > audioresample > > > > ! pulsesink device=MySink > > > > > > > > 3rd shell: > > > > $ gst-launch-1.0 pulsesrc device=MySink.monitor ! audio/x- > > > > raw,channels=2 ! audioconvert ! audioresample ! opusenc ! > > > > oggmux ! > > > > shout2send ip=... port=... mount=... password=... > > > > > > > > After 20 minutes, the sound becomes choppy and stop 4/5 minutes > > > > after. > > > > > > > > You can notice that I don't use the second null sink. > > > > > > > > Is it a normal behavior ? If so, how can I do to achieve this > > > > configuration on my RPi (by using multi null sink) : > > > > > > > > gst_input1 => MySink => gst output1 > > > > gst_input2 => MySink2 => gst output2 > > > > > > > > without any trouble ? > > > > Thanks. > > > > ++ > > > > > > > > Jack > > > > > > > > > > > > > > > > Le vendredi 11 février 2022 à 21:33 +0100, corbeille a écrit : > > > > > Hello, > > > > > > > > > > I am using pulseaudio (version 12.2) to "join" two audio > > > > > streams > > > > > created > > > > > with gstreamer (1.14.4) on a raspberry pi (Raspbian GNU/Linux > > > > > 10 > > > > > (buster)). > > > > > > > > > > Here the real things : > > > > > > > > > > In a first shell, i start with : > > > > > $ pacmd load-module module-null-sink sink_name=MySink > > > > > > > > > > Then I use gstreamer to get the sound from my audio input (in > > > > > an > > > > > other > > > > > shell) : > > > > > $ gst-launch-1.0 pulsesrc > > > > > device="alsa_input.usb-0d8c_C-Media_USB_Headphone_Set- > > > > > 00.analog- > > > > > mono" > > > > > ! > > > > > audioconvert ! audioresample ! pulsesink device=MySink > > > > > > > > > > and I send this sound on an icecast server (in an other > > > > > shell) : > > > > > $ gst-launch-1.0 pulsesrc device=MySink.monitor ! > > > > > "audio/x-raw,channels=2" ! audioconvert ! audioresample ! > > > > > opusenc > > > > > ! > > > > > oggmux ! shout2send ip=... port=... mount=... password=... > > > > > > > > > > Everything is fine, but after 10 minutes, I get in the 3rd > > > > > shell > > > > > ($ > > > > > gst-launch-1.0 pulsesrc device=MySink.monitor ! ...) a > > > > > sequence > > > > > of > > > > > messages like : > > > > > > > > > > WARNING: from element > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Can't record audio fast enough > > > > > Additional debug info: > > > > > gstaudiobasesrc.c(849): gst_audio_base_src_create (): > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Dropped 10080 samples. This is most likely because downstream > > > > > can't > > > > > keep > > > > > up and is consuming samples too slowly. > > > > > WARNING: from element > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Can't record audio fast enough > > > > > Additional debug info: > > > > > gstaudiobasesrc.c(849): gst_audio_base_src_create (): > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Dropped 34560 samples. This is most likely because downstream > > > > > can't > > > > > keep > > > > > up and is consuming samples too slowly. > > > > > WARNING: from element > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Can't record audio fast enough > > > > > Additional debug info: > > > > > gstaudiobasesrc.c(849): gst_audio_base_src_create (): > > > > > /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: > > > > > Dropped 48000 samples. This is most likely because downstream > > > > > can't > > > > > keep > > > > > up and is consuming samples too slowly. > > > > > ... > > > > > > > > > > The sound become glitchy and stop after 2 or 3 minutes. > > > > > > > > > > According to the message, the stream after "pulsesrc > > > > > device=MySink.monitor" is too slow. But I don't understand > > > > > why. > > > > > > > > > > Do you know where is the problem in this setup and how to > > > > > solve > > > > > it ? > > > > > Thanks ! > > > > > ++ > > > > > > > > > > Jack > > > > > > > > Hello, > > > > > > have you tried with a more recent version of pulseaudio? 12.2 is > > > pretty > > > old. You should use 15.0 or current git. > > > > > > Regards > > > Georg > > >
