[pulseaudio-discuss] Dropped xxx samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

Sun Feb 13 20:26:21 UTC 2022

On 13.02.22 21:02, corbeille wrote:
> 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 ?

Yes, that's 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 Well, 14.2 is at least not completely out of date.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
>>>>