Audio streaming problems

Michiel Konstapel michiel at
Wed Dec 21 10:21:30 UTC 2022

Nice! My experience with gstreamer is that it's all in the "subtle 
differences" so I'm glad I could help :)
WRT the video, you can set the encoder to a super low bit rate so it at 
least doesn't needlessly consume bandwidth.

On Monday 19 December 2022 14:52:47 (+01:00), Joel Lord wrote:

 > Thank you, Michiel, you got it in one.


 > aacparse converts an aac stream of audio to "framed", and clearly that 
was what mpegtsmux needed.  Now that you've pointed that out I see it in 
the documentation, but that was a subtle difference I was missing.


 > The video part works fine, I guess I'm stuck with a static image I 
didn't really want.  Worse things have happened.


 > -Joel


 > On 12/19/2022 3:41 AM, Michiel Konstapel wrote:

 > > On Monday 19 December 2022 03:38:38 (+01:00), Joel Lord via 
gstreamer-devel wrote:

 > >  > I'm trying to take an audio feed from a microphone connected via a 
USB interface and (ideally) stream it directly out via HTTP. I want 
multiple end users to be able to get to the stream and listen to it, real 
time, from their phones. I haven't had any luck finding components that add 
up to that, but I did find how to produce a mpeg4ts stream, so I added a 
static image and have that streaming successfully, I added in a clock to 
have something changing to prove that it was working. But the audio never 
comes through.

 > >  >

 > >  > If I take the mpegtsmux and hlssink off the end of my pipeline and 
replace it with a filesink I can prove that my audio is working fine, but 
when I feed it to mpegtsmux and hlssink it seems to get ignored. If I have 
no image it never creates a second segment or the playlist for the stream, 
so I can't see anything at all. If I include the image the stream works but 
it seems to have no audio. I've tried 3 or 4 different audio formats, it 
doesn't complain that I have a format mismatch and it seems to be working, 
but nothing comes through.

 > >  >

 > >  > Using gstreamer1.0 version 1.18.4 on a Raspberry Pi 4.

 > >  >

 > >  > gst-launch-1.0 -v \

 > >  > mpegtsmux name=mux ! hlssink 
location=/var/www/stream/stream_files/segment%05d.ts target-duration=3 
playlist-location=/var/www/stream/playlist.m3u8 \

 > >  > filesrc location=/home/pi/EKPB.png ! decodebin ! \

 > >  > videoconvert ! video/x-raw,format=I420 ! imagefreeze ! \

 > >  > clockoverlay halignment=right ! x264enc ! mux. \

 > >  > alsasrc ! audioconvert ! audio/x-raw,channels=1 ! \

 > >  > level ! avenc_mp2fixed ! queue max-size-buffers=0 max-size-bytes=0 
max-size-time=1000000000 ! mux.

 > >  >

 > >  > If I take the queue out of the audio path it tosses a few warnings 
about dropping samples (a lot of samples) and gives me no faith that it 
will work. Changing the buffer settings on the queue made no difference.

 > >  >

 > >  > So for anyone who has made it this far, I'm quite open to better 
solutions than I've found OR fixing the one that I have that isn't working. 
I don't actually want anything but the audio feed, and I want it as close 
to 0 latency as I can get.

 > >  >

 > >  > Thanks for your help!

 > >  >

 > > Interesting! hlssink expects video key frames to know when it can 
start a new segment (which has to start with a key frame) so I don't know 
what happens with an audio only stream. At least for getting things 
working, it's probably going to be easiest to indeed have a "dummy" video 
track in there.

 > > I use the following for HLS audio:

 > > ... ! avenc_aac ! aacparse ! mpegtsmux

 > > Maybe give that a go? The aacparse might be adding information needed 
by the muxer or the sink.

 > > For x264enc you may need to specify a keyframe interval 
(key-int-max=<number of frames>) but hlssink will actively ask its upstream 
for key frames so that might not be required. Maybe also specify a 
framerate in your video caps, I don't know what the default is. After 
x264enc I also have a parser:

 > > ... ! x264enc ! h264parse config-interval=-1 ! ...

 > > HTH,

 > > Michiel


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