[Bug 796466] New: failed to link avmux_spdif link with alsasink for ac3 passthrough

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=796466

            Bug ID: 796466
           Summary: failed to link avmux_spdif link with alsasink for ac3
                    passthrough
    Classification: Platform
           Product: GStreamer
           Version: 1.14.0
                OS: Linux
            Status: NEW
          Severity: normal
          Priority: Normal
         Component: gst-libav
          Assignee: gstreamer-bugs at lists.freedesktop.org
          Reporter: lyon.wang at nxp.com
        QA Contact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---

Hi, 
   I was trying to realize ac3 audio passthrough via hdmi by a certain pipeline
with avmux_spdif.

   By pipeline:  
"gst-launch-1.0 filesrc location= 5.1_de.ac3 ! ac3parse ! avmux_spdif !
filesink location= ac3.spdif"
  I managed to dump out the encapsulated ac3 (iec61958) format.

  Then I was trying to link avmux_spdif with alsasink

  Noticing avmux_spdif src caps is "application/x-gst-av-spdif", which alsasink
not supported, so I modified avmux_spdif src caps with "audio/x-ac3,
framed=true"

    However seems I still met problem when trying to output the data via
alsasink

below is some log with GST_DEBUG=audiobasesink:6,2

Setting pipeline to PAUSED ...
0:00:00.182977808 [335m 3646[00m     0x3c8bb190 [33;01mWARN   [00m [00m   
         basesrc gstbasesrc.c:3583:gst_base_src_start_complete:<filesrc0>[00m
pad not activated yet
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstAc3Parse:ac3parse0.GstPad:src: caps = audio/x-ac3,
framed=(boolean)true, rate=(int)48000, channels=(int)6, alignment=(string)frame
/GstPipeline:pipeline0/avmux_spdif:avmux_spdif0.GstPad:audio_0: caps =
audio/x-ac3, framed=(boolean)true, rate=(int)48000, channels=(int)6,
alignment=(string)frame
0:00:00.186149288 [335m 3646[00m     0x3c867280 [32;01mFIXME  [00m [00m   
        basesink
gstbasesink.c:3145:gst_base_sink_default_event:<alsasink0>[00m stream-start
event without group-id. Consider implementing group-id handling in the upstream
elements
0:00:00.186471848 [335m 3646[00m     0x3c867280 [37mDEBUG  [00m [00m      
audiobasesink
gstaudiobasesink.c:1197:gst_audio_base_sink_preroll:<alsasink0>[00m ringbuffer
in wrong state
0:00:00.186508208 [335m 3646[00m     0x3c867280 [33;01mWARN   [00m [00m   
   audiobasesink
gstaudiobasesink.c:1198:gst_audio_base_sink_preroll:<alsasink0>[00m error:
sink not negotiated.
0:00:00.186702488 [335m 3646[00m     0x3c867280 [37mDEBUG  [00m [00m      
audiobasesink
gstaudiobasesink.c:1197:gst_audio_base_sink_preroll:<alsasink0>[00m ringbuffer
in wrong state
ERROR: from element /GstPipeline:pipeline0/GstAlsaSink:alsasink0: The stream is
in the wrong format.
0:00:00.186732608 [335m 3646[00m     0x3c867280 [33;01mWARN   [00m [00m   
   audiobasesink
gstaudiobasesink.c:1198:gst_audio_base_sink_preroll:<alsasink0>[00m error:
sink not negotiated.
Additional debug info:
../../../../git/gst-libs/gst/audio/gstaudiobasesink.c(1198):
gst_audio_base_sink_preroll (): /GstPipeline:pipeline0/GstAlsaSink:alsasink0:
sink not negotiated.
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...

Do anyone meet similar issue when trying to use avmux_spdif to passthrough ac3
via alsasink?

Any hint will be appreciated for this issue

Thanks
Lyon

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