CODEWITHSUNDEEP
bluetoothgstreameraudio-streaming
BHARATH M
BHARATH M

Reputation: 11

Gstreamer "rtpsbcdepay" is not working and returning "<rtpsbcdepay0> Segment with non-TIME format not supported" error

This is my actual pipeline

gst-launch-1.0 --gst-debug=3 fdsrc name=fdsrc ! application/x-rtp,media=audio,encoding=SBC,payload=96,clock-rate=44100 ! rtpsbcdepay ! sbcparse ! sbcdec ! audioconvert ! audio/x-raw,layout=interleaved,format=F32LE,channels=2 ! audioresample quality=2 ! appsink name=appsink caps="audio/x-raw,layout=interleaved,format=F32LE,rate=48000,channels=2"

I didn't get audio data and so I tried to find out which command causes the issue. I have used "filesync" to store the data coming from phone and try to debug step by step.

Here is the command to capture the rtp encoded data

"fdsrc name=fdsrc ! application/x-rtp,media=audio,encoding=SBC,payload=96,clock-rate=44100 !filesink location=/tmp/bt_capture.dat"

With this, I got some audio rtp packets in bt_capture.dat file. I used this file as src and add the commands one by one. Getting error in "rtpsbcdepay"

$ gst-launch-1.0 --gst-debug=3 filesrc location=/media/media/Audio/bt_capture.dat ! application/x-rtp,media=audio,encoding=SBC,payload=96,clock-rate=44100,format=time ! rtpsbcdepay ! sbcparse ! sbcdec ! audioconvert ! audio/x-raw,layout=interleaved,format=F32LE,channels=2 ! audioresample quality=2 ! appsink name=appsink caps="audio/x-raw,layout=interleaved,format=F32LE,rate=48000,channels=2" 0:00:00.000177132 19448 0x197d800 INFO GST_INIT gst.c:586:init_pre: Initializing GStreamer Core Library version 1.18.4 0:00:00.000618217 19448 0x197d800 INFO GST_INIT gst.c:587:init_pre: Using library installed in /usr/lib 0:00:00.000884197 19448 0x197d800 INFO GST_INIT gst.c:605:init_pre: Linux W-4B17-BM 5.15.21-rt30-qsc+ #1 SMP PREEMPT_RT Thu Oct 20 12:57:18 UTC 2022 x86_64 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... 0:00:00.062258692 19448 0x1b5cd20 ERROR rtpbasedepayload gstrtpbasedepayload.c:662:gst_rtp_base_depayload_handle_event: Segment with non-TIME format not supported 0:00:00.062676736 19448 0x1b5cd20 ERROR rtpbasedepayload gstrtpbasedepayload.c:662:gst_rtp_base_depayload_handle_event: Segment with non-TIME format not supported 0:00:00.064241297 19448 0x1b5cd20 WARN basesrc gstbasesrc.c:3127:gst_base_src_loop: error: Internal data stream error. 0:00:00.064581497 19448 0x1b5cd20 WARN basesrc gstbasesrc.c:3127:gst_base_src_loop: error: streaming stopped, reason error (-5) ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data stream error. Additional debug info: ../../unpacked/gstreamer-1.18.4/libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc0: streaming stopped, reason error (-5) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ...

Can any one please suggest, how to overcome this error? P.S : I have used format=time, but it didn't help and threw some syntax error

Upvotes: 1

Views: 332

Answers (1)

BHARATH M
BHARATH M

Reputation: 11

Adding a ""rtpjitterbuffer" along with "do-timestamp=TRUE" in fdsrc solves the problem but with some warnings. Still, this warning didn't affect playback (may be some packets missing?)

===================== :01:21.926788667 20721 0x7f47100072a0 WARN audioresample gstaudioresample.c:729:gst_audio_resample_check_discont: encountered timestamp discontinuity of 1874 samples = 0:00:00.042494331 0:01:21.927975600 20721 0x7f47100072a0 WARN audioresample gstaudioresample.c:729:gst_audio_resample_check_discont: encountered timestamp discontinuity of 7968 samples = 0:00:00.180680272 0:01:22.122851228 20721 0x7f47100072a0 WARN audioresample gstaudioresample.c:729:gst_audio_resample_check_discont: encountered timestamp discontinuity of 1874 samples = 0:00:00.042494331

Pipeline:

"fdsrc name=fdsrc do-timestamp=TRUE ! " "application/x-rtp,media=audio,encoding=SBC,payload=96,clock-rate=44100 ! " "rtpjitterbuffer !" "rtpsbcdepay ! " "sbcparse ! " "sbcdec ! " "audioconvert ! " "audio/x-raw,layout=interleaved,format=F32LE,channels=2 ! " "audioresample quality=2 ! " "appsink name=appsink caps="audio/x-raw,layout=interleaved,format=F32LE,rate=48000,channels=2"

Upvotes: 0

Related Questions