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Fwd: help : Regarding AMR-WB in gstreamer.

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piyush

Fwd: help : Regarding AMR-WB in gstreamer.

Respected sir

I hope you are doing well and in good health'

I have two issues:

ISSUE1 -
I am using Kurento Media Server which in turn uses GStreamer.

in our case of Transcoding from PCMA to AMR-WB, we hear the audio for a starting time of 5 seconds, and then a total silence is observed.

I am getting a warning message in logs

0:00:27.154301569  4831 0x14d704002720 WARN         rtpsynchronizer kmsrtpsynchronizer.c:561:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> [Sorted mode] Fix PTS not increasing monotonically, SSRC: 3037524492, seq: 526, rtp_ts: 127015436, ext_ts: 127015436, last: 0:00:13.743311941, current: 0:00:07.723311941, fixed = last: 0:00:13.743311941
0:00:27.154397304  4831 0x14d704002720 WARN                kmsutils kmsutils.c:1479:kms_utils_depayloader_adjust_pts_out:<rtppcmudepay0> Fix PTS not strictly increasing, last: 0:00:13.755311941, current: 0:00:13.743311941, fixed = last + 1: 0:00:13.756311941
0:00:27.174221849  4831 0x14d704002720 LOG          rtpsynchronizer kmsrtpsynchronizer.c:425:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> RTP SSRC: 3037524492, Seq: 527
0:00:27.174277267  4831 0x14d704002720 WARN         rtpsynchronizer kmsrtpsynchronizer.c:549:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> Function Name :kms_rtp_synchronizer_process_rtp_buffer_mapped  File Name : /root/PIYUSH/kms-omni-build/kms-core/src/gst-plugins/commons/kmsrtpsynchronizer.c  Line Number: 549
0:00:27.174299732  4831 0x14d704002720 WARN         rtpsynchronizer kmsrtpsynchronizer.c:561:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> [Sorted mode] Fix PTS not increasing monotonically, SSRC: 3037524492, seq: 527, rtp_ts: 127015596, ext_ts: 127015596, last: 0:00:13.743311941, current: 0:00:07.743311941, fixed = last: 0:00:13.743311941
0:00:27.174394041  4831 0x14d704002720 WARN                kmsutils kmsutils.c:1479:kms_utils_depayloader_adjust_pts_out:<rtppcmudepay0> Fix PTS not strictly increasing, last: 0:00:13.756311941, current: 0:00:13.743311941, fixed = last + 1: 0:00:13.757311941
0:00:27.174525946  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.194253540  4831 0x14d704002720 LOG          rtpsynchronizer kmsrtpsynchronizer.c:425:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> RTP SSRC: 3037524492, Seq: 528
0:00:27.194325675  4831 0x14d704002720 WARN         rtpsynchronizer kmsrtpsynchronizer.c:549:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> Function Name :kms_rtp_synchronizer_process_rtp_buffer_mapped  File Name : /root/PIYUSH/kms-omni-build/kms-core/src/gst-plugins/commons/kmsrtpsynchronizer.c  Line Number: 549
0:00:27.194360607  4831 0x14d704002720 WARN         rtpsynchronizer kmsrtpsynchronizer.c:561:kms_rtp_synchronizer_process_rtp_buffer_mapped:<KmsRtpSynchronizer@0x14d72c020d80> [Sorted mode] Fix PTS not increasing monotonically, SSRC: 3037524492, seq: 528, rtp_ts: 127015756, ext_ts: 127015756, last: 0:00:13.743311941, current: 0:00:07.763311941, fixed = last: 0:00:13.743311941

I am not getting any Error messages and just a print of timestamp discontinuity in the interval of RTP packets

Something like this

mple0> encountered timestamp discontinuity of 960 samples = 0:00:00.060000000
0:00:26.934555639  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:26.974570708  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.014521594  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.054561629  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.094575209  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.122462722  4831 0x14d7100018f0 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample0> encountered timestamp discontinuity of 640 samples = 0:00:00.040000000
0:00:27.134523890  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.174525946  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.214529149  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000
0:00:27.254541540  4831 0x14d704002720 WARN           audioresample gstaudioresample.c:1009:gst_audio_resample_check_discont:<audioresample1> encountered timestamp discontinuity of 304 samples = 0:00:00.038000000

I just want to ask whether is this a bug at GStreamer?

I have reason to believe so because the source(Telenet HardPhone) is sending the correct packets as per RFC verified by me Twice.

I bug given below was for Opus, Maybe there is a chance that it can come for AMR-WB. This issue is arising out of only one type of phone and for rest other phones, the transcoding is working smoothly.

https://bugzilla.gnome.org/show_bug.cgi?id=726579

The above link talks about the same issue regarding the timestamp discontinuity but in the case of Opus Codec. Can this be the same for AMR-WB codec?

Just want your feedback.

This was completely random and difficult to produce at that time, maybe that is the reason people have avoided it?

I am happy to assist with any logs that you may require for the same.

ISSUE 2 - Does Gstreamer support AMR-WB Bandwidth Efficient. I am using GStreamer-1.5.

when i did :

gst-inspect-1.5 rtpamrdepay

Factory Details:
Rank secondary (128)
Long-name RTP AMR depayloader
Klass Codec/Depayloader/Network/RTP
Description Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)
Author Wim Taymans <[hidden email]>

Plugin Details:
Name rtp
Description Real-time protocol plugins
Filename /usr/lib/x86_64-linux-gnu/gstreamer-1.5/libgstrtp.so
Version 1.8.1.1
License LGPL
Source module gst-plugins-good
Source release date 2020-07-23 12:21 (UTC)
Binary package GStreamer Good Plugins (Ubuntu)
Origin URL https://launchpad.net/distros/ubuntu/+source/gst-plugins-good1.5

GObject
+----GInitiallyUnowned
+----GstObject
+----GstElement
+----GstRTPBaseDepayload
+----GstRtpAMRDepay

Pad Templates:
SRC template: 'src'
Availability: Always
Capabilities:
audio/AMR
channels: 1
rate: 8000
audio/AMR-WB
channels: 1
rate: 16000

SINK template: 'sink'
Availability: Always
Capabilities:
application/x-rtp
media: audio
clock-rate: 8000
encoding-name: AMR
octet-align: 1
application/x-rtp
media: audio
clock-rate: 16000
encoding-name: AMR-WB
octet-align: 1

Element Flags:
no flags set

Element Implementation:
Has change_state() function: 0x148b1db19190

Element has no clocking capabilities.
Element has no URI handling capabilities.

Pads:
SINK: 'sink'
Pad Template: 'sink'
SRC: 'src'
Pad Template: 'src'

Element Properties:
name : The name of the object
flags: readable, writable
String. Default: "rtpamrdepay0"
parent : The parent of the object
flags: readable, writable
Object of type "GstObject"
stats : Various statistics
flags: readable
Boxed pointer of type "GstStructure" clock_rate: 0
npt-start: 0
npt-stop: 18446744073709551615
play-speed: 1
play-scale: 1
running-time-dts: 18446744073709551615
running-time-pts: 18446744073709551615
seqnum: 0
timestamp: 0

Also, in the documentation, there was no source of gstreamer supporting AMR-WB Bandwidth Efficient (Octet-Align = 0). Please let me know whether it is supported or not.

Please help us.

Thanks

Piyush Badkul [Personal Website]

+91 8989412425 | +91 7987549194

Designer | Developer (App Developer & Software Developer) | Blogger.

Research Engineer at C-DOT | Co-Founder (YoCount) | Curator (CodesMyth)

Bachelor of Technology, Maulana Azad National Institute Of Technology, Bhopal, India [An Institute of National Importance].

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