Internal data stream error due to stream position with interleave and audiomixer

[Nicolas Dufresne]

Le ven. 19 juill. 2019 15 h 40, <mark.wattier at eqware.net> a écrit :

> I am developing a gstreamer app in C that receives multiple RTP audio
> streams over UDP, sends a copy of each to a WAV file, and sends another
> copy to an alsasrc. To select which stream goes to alsasrc, I am using an
> audiomixer element and I set the volume on the sink pads appropriately.
> This all works when I have a single channel going to alsasrc.
>
> I need to have two streams going to alsasrc, each of which can be selected
> independently. For this, I have set up an interleave element and a second
> audiomixer. Each of the 8 incoming streams has a tee, sending the output to
> each audiomixer.
>
> This also works, but only when I use a single stream for input. When I add
> a second stream, I get an Internal data stream error, reason:
> not-negotiated. With debug on, I chased this back to the audiomixer
> element, where I got the following message:
> 0:00:15.634352592 ^[[332m 1297^[[00m   0x243920 ^[[37mDEBUG  ^[[00m
> ^[[00m          audiomixer
> gstaudiomixer.c:402:gst_audiomixer_setcaps:<audiomixer0:sink_1>^[[00m got
> input caps
> audio/x-raw, layout=(string)interleaved, rate=(int)16000,
> format=(string)S16LE, channels=(int)1,
> channel-mask=(bitmask)0x0000000000000001, but current caps are
> audio/x-raw, layout=(string)interleaved, rate=(int)16000,
> format=(string)S16LE, channels=(int)1,
> channel-mask=(bitmask)0x0000000000000001
>
> The caps are the same. I dug deeper, adding my own GST_DEBUG statements
> and found that gst_audio_info_is_equal was returning false to
> gst_audiomixer_setcaps, because of this section:
> if (memcmp (info->position, other->position,
>           GST_AUDIO_INFO_CHANNELS (info) * sizeof
> (GstAudioChannelPosition)) !=
>       0)
>     return FALSE;
>
> I have tried many combinations of properties for the audiomixer and
> interleave elements. I admit I do not completely understand how to use the
> start-time or alignment threshold properties, but they seem like that could
> lead to a solution.
>
> I am stuck using gstreamer 1.12.2 for this embedded product, using yocto,
> branch rocko. I have a hunch that 1.14.4, from yocto branch thud, might
> resolve this problem, but I would rather not mix up versions in our SDK if
> I can help it, and it can be difficult to port versions across yocto
> branches.
>

I do the same in a more complicated way, I'm pretty sure I can help. I'm on
1.10.3 still for that application, but this didn't work so well without
patches, which I believe most are covered in 1.14+.

The difference between audiomixer and audiointerleaver in such case, is
that adding interleaver pads changes the caps. The other issue is that you
need a channel mapping, I have never figured out how to do that from
gst-launch-1.0.



> The following is a simplified gst-launch pipeline which results in a very
> similar failure. In this example, the error propogates back to the udpsrc
> rather than coming up in audiomixer.
>
> gst-launch-1.0 -v interleave name=i ! \
>  audioconvert ! \
>  alsasink sync=false buffer-time=1000  \
> \
> udpsrc port=5000 buffer-size=1000 ! \
>  "application/x-rtp, media=(string)audio, clock-rate=(int)16000,
> encoding-name=(string)L16" ! \
>  rtpL16depay ! \
>  queue ! \
>  audio/x-raw,format=S16BE,layout=interleaved,rate=16000,channels=1 ! \
>  audioconvert ! \
>  "audio/x-raw,channels=1,channel-mask=(bitmask)0x1" ! \
>  tee name=t0 ! queue ! audiomixer name=mix0 !\
>  i.sink_0  \
> \
> udpsrc port=5001 buffer-size=1000 ! \
>  "application/x-rtp, media=(string)audio, clock-rate=(int)16000,
> encoding-name=(string)L16" ! \
>  rtpL16depay ! \
>  queue ! \
>  audio/x-raw,format=S16BE,layout=interleaved,rate=16000,channels=1 ! \
>  audioconvert ! \
> "audio/x-raw,channels=1,channel-mask=(bitmask)0x2" ! \
>  tee name=t1 ! queue ! audiomixer name=mix1 !\
>  i.sink_1 \
> t0. ! queue ! mix1. \
> t1. ! queue ! mix0.
>
>
>
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