gstreamer-devel Digest, Vol 113, Issue 9

Sat Jun 6 05:32:00 UTC 2020

Re: appsrc + x11 + GLMemory

The more I study the pipeline: gst-inspect-1.0 appsrc stream-type=0
do-timestamp=1emit-signals=0 format=3 is-live=1 name=source
caps="video/x-raw(memory:GLMemory), width=512, height=512,
framerate=(fraction)10/1, format=(string)RGBA,
texture-target=(string)external-oes" ! glcolorconvert ! gldownload !
x264enc tune=0x4 b-adapt=0 ! h264parse ! matroskamux ! filesink
location="/tmp/test.mkv" sync=0"
the more I come to the conclusion that the 'appsrc' element is not good
choice as source for OpenGL's texture. Or gstreamer is not suitable for
these purposes. Really nobody knows about such a feature?

 <gstreamer-devel-request at lists.freedesktop.org>:

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> Today's Topics:
>
>    1. Re: bounties for OpenPOWER optimization (Daniel Pocock)
>    2. Re: Echo cancelling with webrtcdsp (mwon)
>    3. Alexa SDK with gstreamer not playing audio (deeps8us)
>    4. Re: webrtcbin recommandations for Safari iOS (audio only) (Jack)
>    5. Re: Mic on Raspberry Pi - no output from Gstreamer (kotaro)
>    6. Re: Mic on Raspberry Pi - no output from Gstreamer (kotaro)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 4 Jun 2020 14:58:28 +0200
> From: Daniel Pocock <daniel at pocock.pro>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: bounties for OpenPOWER optimization
> Message-ID: <25e96a8b-5ff8-3787-0822-9dd1005de403 at pocock.pro>
> Content-Type: text/plain; charset=utf-8
>
>
>
> On 04/06/2020 09:07, Sebastian Dr?ge wrote:
> > On Thu, 2020-06-04 at 08:32 +0200, Daniel Pocock wrote:
> >>
> >> Are there any improvements to gstreamer and dependencies that could
> >> be funded this way?
> >
> > The audioresample code in gst-plugins-base has some custom assembly
> > that could use a Power implementation. Apart from that improvements to
> > ORC in that regard would be useful and would automatically benefit all
> > GStreamer code using ORC (including video conversion/scaling).
>
> Thanks, those are great tips
>
> I made brief feature requests for those things and submitted them
> through IBM's form.
>
> If you have any ideas to help bounty-hunters get started, you could
> include them in the issues or link them to other issues.
>
> https://gitlab.freedesktop.org/gstreamer/orc/-/issues/29
>
> https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/765
>
> Regards,
>
> Daniel
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 4 Jun 2020 19:15:07 -0500 (CDT)
> From: mwon <miguelwon at gmail.com>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: Echo cancelling with webrtcdsp
> Message-ID: <1591316107258-0.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
> Sorry, only today I was able to return to this problem.
>
> So, the thing is that the webrtcdsp must be placed in the initial pipeline
> ("pipe1") because it is the mic from the alsasrc that must "turned of",
> i.e., the sound that it captures must be filtered out. These are the full
> steps:
>
>
> Remote audio input -> decodebin -> speakers -> mic (alsasrc) -> encode ->
> remote audio output
>
> with decodebin stop only established after the webrtc connection.
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 4 Jun 2020 22:59:31 -0500 (CDT)
> From: deeps8us <deeps8us at gmail.com>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Alexa SDK with gstreamer not playing audio
> Message-ID: <1591329571739-0.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
> We are integrating Alexa SDK on Broadcom platform.
> For audio output, Alexa uses appsrc -> decoder -> decodedQueue -> converter
> -> volume -> audioSink pipeline
>
> https://github.com/alexa/avs-device-sdk/blob/master/MediaPlayer/GStreamerMediaPlayer/include/MediaPlayer/MediaPlayer.h#L159
>
> In a Broadcom device, this gave error and no audio playing
> WARN               decodebin
> gstdecodebin2.c:4679:gst_decode_bin_expose:<decoder> error: no suitable
> plugins found:
> Missing decoder: audio/x-brcm-native (audio/x-brcm-native,
> format=(string)brcm)
>
> gst-launch filesrc location=<mp3_file> ! decodebin ==> also gives same
> error
>
> But instead of decodebin, if I pick individual elements, it work.
> gst-launch filesrc location=<mpe_file> ! mpegaudioparse ! brcmaudiodecoder
> !
> brcmaudiosink (works)
> These same elements are used in decodebin as well.
>
> Any pointers would help.
>
> Thanks
>
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
>
>
> ------------------------------
>
> Message: 4
> Date: Fri, 5 Jun 2020 10:05:31 +0200
> From: Jack <jack at rybn.org>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: webrtcbin recommandations for Safari iOS (audio only)
> Message-ID: <8e9ee3a3-0c18-9fc9-76f6-8163977d5b3b at rybn.org>
> Content-Type: text/plain; charset=utf-8
>
> Hello,
>
> Well, after more tests with Safari iOS. The results are (and I don't
> understand why) :
> - with 4G : KO
> - with wifi : OK
>
> So my app with iOS Safari seems to work with webrtcbin, super !
> Now, if someone can explain why it is not working on 4G ?
> With other browser on 4G, it is working fine.
> Best!
> ++
>
> Jack
>
>
>
> Le 26/05/2020 ? 21:14, Jack a ?crit?:
> > I can add to the working list :
> > - MacOSX with Firefox, Chrome and Safari
> >
> > The javascript is visible on (and the nice noise can be listen on) :
> > https://discrepant.me/
> >
> > So, why it is not working on Safari iOS ? Something to
> > do/detect/add/check/modify ? Any recommandations ?
> > Thanx.
> >
> > GStreamer 1.16.1
> >
> > ++
> >
> > Jack
> >
> >
> >
> > Le 24/05/2020 ? 23:49, Jack a ?crit?:
> >> Hi list !
> >>
> >> Do you have any recommandations to make webrtcbin working with Safari on
> >> iOS ? The idea is to get (only) the audio stream in the browser.
> >>
> >> I am asking this because my gstreamer/javascript configuration doesn't
> >> work with Safari iOS.
> >>
> >> However, it is working fine on :
> >> - Linux with Firefox, Chromium and Chrome
> >> - Windows with Firefox
> >> - Android with Firefox and Chrome
> >>
> >>
> >> Here my pipeline using audiotestsrc :
> >> webrtcbin name=sendrecv bundle-policy=max-bundle audiotestsrc
> >> is-live=true wave=red-noise ! audioconvert ! audioresample ! queue !
> >> opusenc ! rtpopuspay ! queue !
> >> application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
> >>
> >> Here the web page to test :
> >> https://discrepant.me/
> >>
> >> Any hint is very welcome !
> >> ++
> >>
> >> Jack
> >>
> >> _______________________________________________
> >> gstreamer-devel mailing list
> >> gstreamer-devel at lists.freedesktop.org
> >> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> >>
> >
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.freedesktop.org
> > https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
> >
>
>
>
> ------------------------------
>
> Message: 5
> Date: Fri, 5 Jun 2020 04:26:38 -0500 (CDT)
> From: kotaro <kotaro-chan at hotmail.de>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: Mic on Raspberry Pi - no output from Gstreamer
> Message-ID: <1591349198698-0.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
> Hello,
>
> i facing same Problem,
> i try to use I2S mems Microphone on Raspi-Zero
>
> gst-launch-1.0 alsasrc device=plughw:0,0 !
> audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=1 ! volume
> volume=10 ! volume volume=3 ! opusenc ! oggmux ! filesink location=test.ogg
>
> does not work.
> i use arecord -l and get:
> card 0: sndrpisimplecar [snd_rpi_simple_card], device 0:
> simple-card_codec_link snd-soc-dummy-dai-0 []
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
>
> i get same problem whileusing arecord to a wave file i get an Sound, so i
> think that i2s MEMS work probertly.
> But when using gst-launch-1.0 alsasrc i get only silent.
>
> Did anyone clear these problem?
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
>
>
> ------------------------------
>
> Message: 6
> Date: Fri, 5 Jun 2020 04:24:05 -0500 (CDT)
> From: kotaro <kotaro-chan at hotmail.de>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Re: Mic on Raspberry Pi - no output from Gstreamer
> Message-ID: <1591349045606-0.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
> Hello,
>
> i facing same Problem,
> i try to use I2S mems Microphone on Raspi-Zero
>
> gst-launch-1.0 alsasrc device=plughw:0,0 !
> audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=1 ! volume
> volume=10 ! volume volume=3 ! opusenc ! oggmux ! filesink location=test.ogg
>
> does not work.
> i use arecord -l and get:
> card 0: sndrpisimplecar [snd_rpi_simple_card], device 0:
> simple-card_codec_link snd-soc-dummy-dai-0 []
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0
>
> i get same problem whileusing arecord to a wave file i get an Sound, so i
> think that i2s MEMS work probertly.
> But when using gst-launch-1.0 alsasrc i get only silent.
>
> Did anyone clear these problem?
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
>
>
> ------------------------------
>
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>
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> ------------------------------
>
> End of gstreamer-devel Digest, Vol 113, Issue 9
> ***********************************************
>
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