Fwd: help : Regarding AMR-WB in gstreamer.
PIYUSH BADKUL
piyushmact18 at gmail.com
Thu Aug 6 16:17:40 UTC 2020
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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 <wim.taymans at gmail.com>
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 <http://opg7371.github.io/>]
*+91 8989412425 |* *+91 7987549194*
Designer | Developer (App Developer & Software Developer) | Blogger.
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Bachelor of Technology, Maulana Azad National Institute Of Technology,
Bhopal, India [An Institute of National Importance].
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