<div dir="ltr"><div class="gmail_default" style="font-size:small">Respected sir </div><div class="gmail_quote"><div dir="ltr"><div style="font-size:small"> </div><div style="font-size:small">I hope you are doing well and in good health'</div><div style="font-size:small">I have two issues: </div><div style="font-size:small"> </div><div style="font-size:small">ISSUE1 - I am using Kurento Media Server which in turn uses GStreamer.</div> 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.<div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px">I am getting a warning message in logs </div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:1px solid rgb(187,187,187);font-size:13px;background-color:rgb(250,250,250)"><code style="margin:0px;padding:0px;border:0px"><div style="margin:0px;padding:0px;border:0px">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 </div></code></div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> I am not getting any Error messages and just a print of timestamp discontinuity in the interval of RTP packets</div><div style="margin:0px;padding:0px;border:0px;font-size:13px">Something like this </div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px"><div style="margin:0px;padding:0px;border:1px solid rgb(187,187,187);background-color:rgb(250,250,250)"><code style="margin:0px;padding:0px;border:0px"><div style="margin:0px;padding:0px;border:0px"><div style="margin:0px;padding:0px;border:0px">mple0> encountered timestamp discontinuity of 960 samples = 0:00:00.060000000</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div><div style="margin:0px;padding:0px;border:0px">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</div></div></code></div><div style="margin:0px;padding:0px;border:0px"> </div>I just want to ask whether is this a bug at GStreamer?</div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px">I have reason to believe so because the source(Telenet HardPhone) is sending the correct packets as per RFC verified by me Twice.</div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px"><div style="font-size:small">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.</div> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px"><a href="https://bugzilla.gnome.org/show_bug.cgi?id=726579" rel="nofollow" style="margin:0px;padding:0px;border:0px;text-decoration-line:none;color:rgb(102,17,204)" target="_blank">https://bugzilla.gnome.org/show_bug.cgi?id=726579</a></div><div style="margin:0px;padding:0px;border:0px;font-size:13px"> </div><div style="margin:0px;padding:0px;border:0px;font-size:13px">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?</div><div style="margin:0px;padding:0px;border:0px;font-size:13px">Just want your feedback.</div><div style="font-size:small">This was completely random and difficult to produce at that time, maybe that is the reason people have avoided it? </div><div><div dir="ltr" data-smartmail="gmail_signature"><div dir="ltr"><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr">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 <<a href="mailto:wim.taymans@gmail.com" target="_blank">wim.taymans@gmail.com</a>> 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 <a href="https://launchpad.net/distros/ubuntu/+source/gst-plugins-good1.5" target="_blank">https://launchpad.net/distros/ubuntu/+source/gst-plugins-good1.5</a> 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</div></div></div></div></div></div></div></div></div></div></div><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr">Piyush Badkul [<a href="http://opg7371.github.io/" style="font-family:Arial,sans-serif;font-size:10px" target="_blank">Personal Website</a>]+91 8989412425 | +91 7987549194Designer | Developer (App Developer & Software Developer) | Blogger. Research Engineer at <a href="http://www.cdot.in/home.htm" target="_blank">C-DOT</a> | Co-Founder (<a href="https://play.google.com/store/apps/details?id=org.shpstartup.android.yocount" target="_blank">YoCount</a>) | Curator (<a href="http://www.codesmyth.com/" style="font-weight:bold" target="_blank">CodesMyth</a>) Bachelor of Technology, Maulana Azad National Institute Of Technology, Bhopal, India [An Institute of National Importance].■ Android ■ Designer ■ Blogger ■ DeveloperNote: This email and its attachments are confidential and are protected by copyright. When addressed to our clients it is subject to our terms and conditions of business. Unless specifically specified, the contents are for information and discussion only </div></div></div></div></div></div></div></div></div></div></div></div></div></div></div></div> </div></div>