I have one audio AAC RTSP stream that I'm trying to pitch it before
outputting to WebRTC.
I have the following pipe that works (RTSP to WebRTC) - (with no pitch):
rtspsrc location=rtsp://user:pass@192.168.6.240:7011/camxxx user-id=user
user-pw=pass latency=100 name=demuxer2 is-live=true protocols=udp
rfc7273-sync=true ! decodebin ! audioconvert ! audioresample ! opusenc !
rtpopuspay timestamp-offset=0 !
application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! tee
name=camtee ! fakesink
I also have the following pitch cmd-line that works on my sound card(distort
the sound in higher tones):
gst-launch-1.0 rtspsrc
location=location=rtsp://user:pass@192.168.6.240:7011/camxxx user-id=user
user-pw=pass latency=100 is-live=true protocols=udp rfc7273-sync=true !
decodebin ! audioconvert ! audioresample ! pitch pitch=1.6 !
autoaudiosink
but when I add the pitch to the first pipe:
rtspsrc location=rtsp://user:pass@192.168.6.240:7011/camxxx user-id=user
user-pw=pass latency=100 name=demuxer2 is-live=true protocols=udp
rfc7273-sync=true ! decodebin ! audioconvert ! audioresample ! pitch
pitch=1.6 ! audioconvert ! opusenc ! rtpopuspay timestamp-offset=0 !
application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! tee
name=camtee ! fakesink
I get "The stream is in the wrong format".
** (xxxxx.exe:10688): CRITICAL **: 21:13:26.789: gst_audio_buffer_map:
assertion '(!meta && info->layout == GST_AUDIO_LAYOUT_INTERLEAVED) || (meta
&& info->layout == meta->info.layout)' failed
Error received from element audioconvert3: The stream is in the wrong
format.
Do you have any idea how can I pitch (distort) an audio RTSP stream and
output OPUS to webRTC?
Thanks,
Mihai
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