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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?
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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 <[hidden email]>
To: [hidden email]
Subject: Re: bounties for OpenPOWER optimization
Message-ID: <[hidden email]>
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 <[hidden email]>
To: [hidden email]
Subject: Re: Echo cancelling with webrtcdsp
Message-ID: <[hidden email]>
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.
--
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------------------------------
Message: 3
Date: Thu, 4 Jun 2020 22:59:31 -0500 (CDT)
From: deeps8us <[hidden email]>
To: [hidden email]
Subject: Alexa SDK with gstreamer not playing audio
Message-ID: <[hidden email]>
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
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------------------------------
Message: 4
Date: Fri, 5 Jun 2020 10:05:31 +0200
From: Jack <[hidden email]>
To: [hidden email]
Subject: Re: webrtcbin recommandations for Safari iOS (audio only)
Message-ID: <[hidden email]>
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
>> [hidden email]
>> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>>
>
> _______________________________________________
> gstreamer-devel mailing list
> [hidden email]
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
------------------------------
Message: 5
Date: Fri, 5 Jun 2020 04:26:38 -0500 (CDT)
From: kotaro <[hidden email]>
To: [hidden email]
Subject: Re: Mic on Raspberry Pi - no output from Gstreamer
Message-ID: <[hidden email]>
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?
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------------------------------
Message: 6
Date: Fri, 5 Jun 2020 04:24:05 -0500 (CDT)
From: kotaro <[hidden email]>
To: [hidden email]
Subject: Re: Mic on Raspberry Pi - no output from Gstreamer
Message-ID: <[hidden email]>
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?
--
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