Audio streaming problems

Joel Lord jpl at
Mon Dec 19 13:52:47 UTC 2022

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.


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 
> playlist-root=http://stream.ek:80/stream_files 
> 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

Joel Lord

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