Capturing and encoding live video and audio: AMD FX(tm)-8350 Eight-Core Processor not fast enough?

[Rodrigo Severo]

Hi,

I'm trying to use gstreamer to capture and encode live video and audio from a USB dongle.

My last attempt:

gst-launch-1.0 v4l2src device=/dev/video0 norm=45056 ! queue ! videoconvert ! videoscale ! videorate ! 'video/x-raw, width=640, height=400, framerate=15/1, format=I420' ! x264enc ! h264parse ! stream. pulsesrc device="alsa_input.usb-Hauppauge_Hauppauge_Device_0013745411-01.analog-stereo" ! queue ! audioconvert ! audioresample ! 'audio/x-raw, channels=2, rate=48000' ! lamemp3enc ! mpegaudioparse ! stream. mpegtsmux name=stream ! queue ! filesink location=/home/rodrigo/temp/test.mpeg

To my surprise I get several messages like:

Setting pipeline to PAUSED ...

Pipeline is live and does not need PREROLL ...

Setting pipeline to PLAYING ...

New clock: GstPulseSrcClock

Redistribute latency...

Redistribute latency...

Redistribute latency...

WARNING: from element /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0: Can't record audio fast enough

Additional debug info:

gstaudiobasesrc.c(847): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0:

Dropped 125760 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

All messages from the warning to the "too slowly" are repeated all the time.

My first surprise: I believe an eight core processor should be much more than enough to do such a job without problems but apparently it isn't.

My second surprise,: it's the _audio_ that's being too slowly processed, not the video.

In reality I believe I'm doing something wrong. Can someone help me find what it is?

BTW, I'm using GStreamer 1.12.3 on Ubuntu 17.10.

Regards,

Rodrigo Severo

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[Nicolas Dufresne-5]

Le jeudi 21 décembre 2017 à 15:35 -0200, Rodrigo Severo a écrit :

> gst-launch-1.0 v4l2src device=/dev/video0 norm=45056 ! queue !
> videoconvert ! videoscale ! videorate ! 'video/x-raw, width=640,
> height=400, framerate=15/1, format=I420' ! x264enc ! h264parse !
> stream. pulsesrc device="alsa_input.usb-
> Hauppauge_Hauppauge_Device_0013745411-01.analog-stereo" ! queue !
> audioconvert ! audioresample ! 'audio/x-raw, channels=2, rate=48000'
> ! lamemp3enc ! mpegaudioparse ! stream. mpegtsmux name=stream ! queue
> ! filesink location=/home/rodrigo/temp/test.mpeg
>
> To my surprise I get several messages like:
>
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstPulseSrcClock
> Redistribute latency...
> Redistribute latency...
> Redistribute latency...
> WARNING: from element /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0:
> Can't record audio fast enough
> Additional debug info:
> gstaudiobasesrc.c(847): gst_audio_base_src_create ():
> /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0:
> Dropped 125760 samples. This is most likely because downstream can't
> keep up and is consuming samples too slowly.
>
> All messages from the warning to the "too slowly" are repeated all
> the time.
>
> My first surprise: I believe an eight core processor should be much
> more than enough to do such a job without problems but apparently it
> isn't.
>
> My second surprise,: it's the _audio_ that's being too slowly
> processed, not the video.
>
> In reality I believe I'm doing something wrong. Can someone help me
> find what it is?
>
> BTW, I'm using GStreamer 1.12.3 on Ubuntu 17.10.

It's propably a buffering issue. Disable bytes size limited on your
queus, maybe that will fix it. Note that x264enc in default
configuration is designed of offline encoding. Just retry and
tune=zerolatency you'll see a large difference.

Nicolas

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[Rodrigo Severo]

2017-12-21 16:48 GMT-02:00 Nicolas Dufresne <[hidden email]>:

Le jeudi 21 décembre 2017 à 15:35 -0200, Rodrigo Severo a écrit :
> gst-launch-1.0 v4l2src device=/dev/video0 norm=45056 ! queue !
> videoconvert ! videoscale ! videorate ! 'video/x-raw, width=640,
> height=400, framerate=15/1, format=I420' ! x264enc ! h264parse !
> stream. pulsesrc device="alsa_input.usb-
> Hauppauge_Hauppauge_Device_0013745411-01.analog-stereo" ! queue !
> audioconvert ! audioresample ! 'audio/x-raw, channels=2, rate=48000'
> ! lamemp3enc ! mpegaudioparse ! stream. mpegtsmux name=stream ! queue
> ! filesink location=/home/rodrigo/temp/test.mpeg
>
> To my surprise I get several messages like:
>
> Setting pipeline to PAUSED ...
> Pipeline is live and does not need PREROLL ...
> Setting pipeline to PLAYING ...
> New clock: GstPulseSrcClock
> Redistribute latency...
> Redistribute latency...
> Redistribute latency...
> WARNING: from element /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0:
> Can't record audio fast enough
> Additional debug info:
> gstaudiobasesrc.c(847): gst_audio_base_src_create ():
> /GstPipeline:pipeline0/GstPulseSrc:pulsesrc0:
> Dropped 125760 samples. This is most likely because downstream can't
> keep up and is consuming samples too slowly.
>
> All messages from the warning to the "too slowly" are repeated all
> the time.
>
> My first surprise: I believe an eight core processor should be much
> more than enough to do such a job without problems but apparently it
> isn't.
>
> My second surprise,: it's the _audio_ that's being too slowly
> processed, not the video.
>
> In reality I believe I'm doing something wrong. Can someone help me
> find what it is?
>
> BTW, I'm using GStreamer 1.12.3 on Ubuntu 17.10.

It's propably a buffering issue. Disable bytes size limited on your
queus, maybe that will fix it. Note that x264enc in default
configuration is designed of offline encoding. Just retry and
tune=zerolatency you'll see a large difference.

Yes, it seems to work much  better.

Thanks,

Rodrigo

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