delay between speaker and microphone
J.Paixao at TELEVIC.com
Thu Jan 28 03:00:59 PST 2016
> -----Original Message-----
> From: gstreamer-devel [mailto:gstreamer-devel-
> bounces at lists.freedesktop.org] On Behalf Of siddarthmamidanna
> Sent: Thursday, January 28, 2016 5:34 AM
> To: gstreamer-devel at lists.freedesktop.org
> Subject: RE: delay between speaker and microphone
> The below pipeline we are using to record at speaker and microphone side:
> gst-launch-1.0 -v filesrc location=/home/pi/record.wav ! audio/x-raw,
> layout=interleaved, rate=8000, format=S16LE, channels=2, endianness=4321,
> width=16, depth=16, signed=true ! tee name=waveaudio ! queue !
> audioconvert ! autoaudiosink waveaudio. ! filesink
> location=record_speaker_8000.wav &
> gst-launch-1.0 -v alsasrc ! audioresample ! audioconvert ! tee
> name=near_audio ! queue ! audio/x-raw, format=S16LE, layout=interleaved,
> rate=8000, channels=2, endianness=4321, width=16, depth=16, signed=true !
> appsink name=soundSink drop=true near_audio. ! filesink
It would be a good idea to set your alsasrc buffer-time if you want to have a bit of control over the latency.
By default the buffer-time is set to 200ms which is already too big for your application.
How do you measure the delay between the microphone and the speaker?
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