# Application Development

**URL:** https://discourse.gstreamer.org/c/app-dev/7.md?page=4

[Latest](https://discourse.gstreamer.org/latest.md) · [Categories](https://discourse.gstreamer.org/categories.md) · [Tags](https://discourse.gstreamer.org/tags.md)

**Page:** 5

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## [Resolution of textoverlay is poor when video is scaled up](https://discourse.gstreamer.org/t/resolution-of-textoverlay-is-poor-when-video-is-scaled-up/5458)

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**Author:** [@andymrob](https://discourse.gstreamer.org/u/andymrob)\
**Replies:** 6\
**Last updated:** [October 11, 2025, 3:20pm UTC](https://discourse.gstreamer.org/t/resolution-of-textoverlay-is-poor-when-video-is-scaled-up/5458 "2025-10-11T15:20:25Z")

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I am programatically displaying a lo-res video file with subtitles from an external srt file. When I drag the edges of the on-screen video window to make it bigger the video becomes blurry as we would expect, but unfortu…

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## [Control one video input when using multiple](https://discourse.gstreamer.org/t/control-one-video-input-when-using-multiple/5443)

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**Author:** [@Birdo](https://discourse.gstreamer.org/u/Birdo)\
**Replies:** 1\
**Last updated:** [October 9, 2025, 5:30pm UTC](https://discourse.gstreamer.org/t/control-one-video-input-when-using-multiple/5443 "2025-10-09T17:30:09Z")

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Hi, I have been struggling with the following issue and I am wondering if I am on the correct path to make it work. I want to create multiple video inputs from a file to be shown on the screen. The video inputs contai…

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## [Running-time in appsrc callback](https://discourse.gstreamer.org/t/running-time-in-appsrc-callback/5444)

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**Author:** [@TheGiamig](https://discourse.gstreamer.org/u/TheGiamig)\
**Replies:** 4\
**Last updated:** [October 9, 2025, 6:52am UTC](https://discourse.gstreamer.org/t/running-time-in-appsrc-callback/5444 "2025-10-09T06:52:49Z")

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Hi there, I would get the running-time in the need-data callback. I can’t use gst\_segment\_to\_running\_time because I don’t known how obtain the segment. May be better if I explain the problem I’m facing: my pipeline ge…

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## [Internal data stream error while streaming to AVTP sink](https://discourse.gstreamer.org/t/internal-data-stream-error-while-streaming-to-avtp-sink/1594)

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**Author:** [@Pranav\_S](https://discourse.gstreamer.org/u/Pranav_S)\
**Replies:** 1\
**Last updated:** [October 8, 2025, 10:13pm UTC](https://discourse.gstreamer.org/t/internal-data-stream-error-while-streaming-to-avtp-sink/1594 "2025-10-08T22:13:54Z")

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Hello, I am trying to stream the video as AVTP packets using video.mp4 file stored in a memory as source. I want to send the AVTP packets at 33 ms interval (considering 30 fps rate). My pipeline is as below, Interface…

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## [Send screencapture data from Pipewire to WebRTC](https://discourse.gstreamer.org/t/send-screencapture-data-from-pipewire-to-webrtc/5426)

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**Author:** [@czscout](https://discourse.gstreamer.org/u/czscout)\
**Replies:** 11\
**Last updated:** [October 8, 2025, 6:23am UTC](https://discourse.gstreamer.org/t/send-screencapture-data-from-pipewire-to-webrtc/5426 "2025-10-08T06:23:29Z")

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I’m attempting to stream a Freedesktop ScreenCast capture over WebRTC via GStreamer using its pipewiresrc and webrtcbin elements. I’ve made a decent amount of progress so far: I can stream the videotestsrc with my curre…

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## [Keep closed captions in the video recording of a baseband signal](https://discourse.gstreamer.org/t/keep-closed-captions-in-the-video-recording-of-a-baseband-signal/5450)

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**Author:** [@lfrdgh](https://discourse.gstreamer.org/u/lfrdgh)\
**Replies:** 0\
**Last updated:** [October 7, 2025, 1:17am UTC](https://discourse.gstreamer.org/t/keep-closed-captions-in-the-video-recording-of-a-baseband-signal/5450 "2025-10-07T01:17:43Z")

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Hi there, I am using a Blackmagic Decklink card on which I have connected a baseband signal to one of its input pins. With GStreamer, I record one-minute videos continuously. However, that baseband signal already carrie…

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## [Attaching a branch to a running pipeline puts it into PAUSED state and omxh264enc doesn't recover](https://discourse.gstreamer.org/t/attaching-a-branch-to-a-running-pipeline-puts-it-into-paused-state-and-omxh264enc-doesnt-recover/1598)

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**Author:** [@rclark](https://discourse.gstreamer.org/u/rclark)\
**Replies:** 2\
**Last updated:** [October 6, 2025, 8:03pm UTC](https://discourse.gstreamer.org/t/attaching-a-branch-to-a-running-pipeline-puts-it-into-paused-state-and-omxh264enc-doesnt-recover/1598 "2025-10-06T20:03:44Z")

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Hello, I’m trying to create a dynamic pipeline that has a trunk that encodes the framebuffer with the omxh264enc element and multiple branches, one for writing to a file and one for sending the encoded data over UDP. The…

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## [decodebin not found on Android with GStreamer 1.24 and NDK 27](https://discourse.gstreamer.org/t/decodebin-not-found-on-android-with-gstreamer-1-24-and-ndk-27/5440)

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**Author:** [@Artemy\_vel](https://discourse.gstreamer.org/u/Artemy_vel)\
**Replies:** 1\
**Last updated:** [October 3, 2025, 12:47am UTC](https://discourse.gstreamer.org/t/decodebin-not-found-on-android-with-gstreamer-1-24-and-ndk-27/5440 "2025-10-03T00:47:01Z")

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Hello, I’m using GStreamer 1.24 on Android with NDK 27. I try to create a decodebin element like this: GstElement \*decodebin = gst\_element\_factory\_make("decodebin", "decodebin"); But I get the error: no such element…

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## [Dynamically link and start/pause a filesrc to a running pipeline with a live source](https://discourse.gstreamer.org/t/dynamically-link-and-start-pause-a-filesrc-to-a-running-pipeline-with-a-live-source/5438)

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**Author:** [@Frank](https://discourse.gstreamer.org/u/Frank)\
**Replies:** 0\
**Last updated:** [October 2, 2025, 8:11am UTC](https://discourse.gstreamer.org/t/dynamically-link-and-start-pause-a-filesrc-to-a-running-pipeline-with-a-live-source/5438 "2025-10-02T08:11:34Z")

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The main branch of my pipeline: audiotestsrc(is-live=true) -\> audioconvert -\> audiomixer(force-live=true) -\> capsfilter -\> queue -\> wasapisink When the pipeline is PLAYING I want to dynamically link the following branc…

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## [SDL interop with Vulkan](https://discourse.gstreamer.org/t/sdl-interop-with-vulkan/5327)

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**Author:** [@anonymix007](https://discourse.gstreamer.org/u/anonymix007)\
**Replies:** 1\
**Last updated:** [October 2, 2025, 11:55am UTC](https://discourse.gstreamer.org/t/sdl-interop-with-vulkan/5327 "2025-10-02T11:55:29Z")

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I’ve attempted to implement a simple demo, however, it only draw the all-time classic black image onto the window. main.c: #define \_GNU\_SOURCE #include \<assert.h\> #include \<stdio.h\> #include \<time.h\> #include \<pthread.…

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## [Gst-plugin-quinn Optimal parameters](https://discourse.gstreamer.org/t/gst-plugin-quinn-optimal-parameters/5431)

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**Author:** [@SamDudley](https://discourse.gstreamer.org/u/SamDudley)\
**Replies:** 3\
**Last updated:** [October 2, 2025, 11:22am UTC](https://discourse.gstreamer.org/t/gst-plugin-quinn-optimal-parameters/5431 "2025-10-02T11:22:59Z")

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I’m looking to get the best performance out of a RTP over Quic stream using quinnquicsink/quinnquicsrc. What are the best parameters to adjust to improve performance?

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## [Application stuck in gst\_init](https://discourse.gstreamer.org/t/application-stuck-in-gst-init/5428)

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**Author:** [@JeffZZ](https://discourse.gstreamer.org/u/JeffZZ)\
**Replies:** 0\
**Last updated:** [September 30, 2025, 2:27pm UTC](https://discourse.gstreamer.org/t/application-stuck-in-gst-init/5428 "2025-09-30T14:27:01Z")

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I have a windows application that is deployed on multiple machines using Windows 11, and it uses some features of Gstreamer, the problem that I have encountered is that, one of these machines gets stuck when the executio…

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## [Change default core library path](https://discourse.gstreamer.org/t/change-default-core-library-path/4544)

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**Author:** [@JeffZZ](https://discourse.gstreamer.org/u/JeffZZ)\
**Replies:** 11\
**Last updated:** [September 30, 2025, 2:20pm UTC](https://discourse.gstreamer.org/t/change-default-core-library-path/4544 "2025-09-30T14:20:57Z")

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Hi, I’m working on Windows, using C++ and Visual Studio. I’ve added an implementation of gstreamer into an already existing project and now I need to pack it and test it into our target machines. The problem is that du…

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## [Question about correct usage of Livesync](https://discourse.gstreamer.org/t/question-about-correct-usage-of-livesync/5328)

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**Author:** [@Kinsi](https://discourse.gstreamer.org/u/Kinsi)\
**Replies:** 4\
**Last updated:** [September 27, 2025, 8:23pm UTC](https://discourse.gstreamer.org/t/question-about-correct-usage-of-livesync/5328 "2025-09-27T20:23:30Z")

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I have a custom Gstreamer Pipeline thats split into two seperate parts, producer and consumer, linked together via Appsrc/Sink The consumer eventually outputs its result via RTMP and needs to stay up even if the produce…

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## [Muxing seperate timestamped h264/5 video and AAC-ELD audio streams](https://discourse.gstreamer.org/t/muxing-seperate-timestamped-h264-5-video-and-aac-eld-audio-streams/5421)

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**Author:** [@fduncanh](https://discourse.gstreamer.org/u/fduncanh)\
**Replies:** 1\
**Last updated:** [September 27, 2025, 5:25pm UTC](https://discourse.gstreamer.org/t/muxing-seperate-timestamped-h264-5-video-and-aac-eld-audio-streams/5421 "2025-09-27T17:25:24Z")

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I am a maintainer of the GPLv3 AirPlay server UxPlay. Apple devices in screen mirror mode send separate encrypted streams of AAC-ELD audio and H264/H265 video, which we decrypt. Each has timestamps (different formats …

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## [Starting a rtsp server via gstreamer on an android device](https://discourse.gstreamer.org/t/starting-a-rtsp-server-via-gstreamer-on-an-android-device/5411)

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**Author:** [@Aren](https://discourse.gstreamer.org/u/Aren)\
**Replies:** 0\
**Last updated:** [September 23, 2025, 10:16am UTC](https://discourse.gstreamer.org/t/starting-a-rtsp-server-via-gstreamer-on-an-android-device/5411 "2025-09-23T10:16:48Z")

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Hello, i am trying to start a rtspserver via gstreamer on android device. I am getting some error messages and i think i have some misconfiguration in android stuio because i get several file not found Errors following …

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## [Issues with mpegtsmux output format](https://discourse.gstreamer.org/t/issues-with-mpegtsmux-output-format/5406)

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**Author:** [@barsigor](https://discourse.gstreamer.org/u/barsigor)\
**Replies:** 0\
**Last updated:** [September 22, 2025, 9:03am UTC](https://discourse.gstreamer.org/t/issues-with-mpegtsmux-output-format/5406 "2025-09-22T09:03:58Z")

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Hello, I’ve a question about gstreamer mpegtsmux plugin. I’m working with Raspberry PI camera i need to provide a mpeg-ts feed with a h264 stream, sent via network, if I use libav the system works (i modified the pipel…

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## [pipeline error for HDMI-RX](https://discourse.gstreamer.org/t/pipeline-error-for-hdmi-rx/5400)

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**Author:** [@N0tiK](https://discourse.gstreamer.org/u/N0tiK)\
**Replies:** 0\
**Last updated:** [September 21, 2025, 12:45am UTC](https://discourse.gstreamer.org/t/pipeline-error-for-hdmi-rx/5400 "2025-09-21T00:45:20Z")

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Im trying to achieve a raw pipeline for hdmi input with a minimum armbian ubuntu image, pipeline works well without kmssink in desktop, but using kmssink in minimum yields errors & timing issues. GST\_VIDEO\_CONVERT\_USE\_R…

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## [audio appsrc using generated bindings](https://discourse.gstreamer.org/t/audio-appsrc-using-generated-bindings/5362)

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**Author:** [@ea7kir](https://discourse.gstreamer.org/u/ea7kir)\
**Replies:** 7\
**Last updated:** [September 17, 2025, 8:16am UTC](https://discourse.gstreamer.org/t/audio-appsrc-using-generated-bindings/5362 "2025-09-17T08:16:33Z")

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I need to push a slice of audio bytes using an appsrc audioSrc, ok := gst.ElementFactoryMake("appsrc", "audioSrc").(gstapp.AppSrc) if !ok { log.Fatalf("failed to create appsrc") } audioInfo := gstaudio.NewAudioInf…

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## [Record video from data received from appsrc (shared memory)](https://discourse.gstreamer.org/t/record-video-from-data-received-from-appsrc-shared-memory/5365)

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**Author:** [@lfrdgh](https://discourse.gstreamer.org/u/lfrdgh)\
**Replies:** 2\
**Last updated:** [September 16, 2025, 2:34pm UTC](https://discourse.gstreamer.org/t/record-video-from-data-received-from-appsrc-shared-memory/5365 "2025-09-16T14:34:36Z")

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Hi everyone, I’m using gstreamer bindings for Golang to record MP4 videos from frame data I receive from shared memory. The data I receive is constantly delivered as a byte array. This data is managed with appsrc. When …

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## [Matroskamux and changes to SPS/PPS](https://discourse.gstreamer.org/t/matroskamux-and-changes-to-sps-pps/5363)

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**Author:** [@jonasdn](https://discourse.gstreamer.org/u/jonasdn)\
**Replies:** 2\
**Last updated:** [September 10, 2025, 2:29pm UTC](https://discourse.gstreamer.org/t/matroskamux-and-changes-to-sps-pps/5363 "2025-09-10T14:29:38Z")

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Hello codec hackers! We had a customer ingesting video to us have a bad experience … it turned out our pipeline with a matroskamux kept erroring out because some values in SPS/PPS kept changing in the input stream. To …

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## [Qmlsink example on imx8qxp board doesn't render videotstsrc (only white screen)](https://discourse.gstreamer.org/t/qmlsink-example-on-imx8qxp-board-doesnt-render-videotstsrc-only-white-screen/4579)

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**Author:** [@roodani](https://discourse.gstreamer.org/u/roodani)\
**Replies:** 4\
**Last updated:** [September 6, 2025, 6:31am UTC](https://discourse.gstreamer.org/t/qmlsink-example-on-imx8qxp-board-doesnt-render-videotstsrc-only-white-screen/4579 "2025-09-06T06:31:06Z")

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Hello, I am trying to launch the qmlsink example on my board, but all I get is a white screen. Any idea how to debug this issue ?, Here is the log of GST\_DEBUG=4 0:00:00.003860125 507 0xaaab187e4660 INFO …

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## [Dynamically removing webrtcsrc](https://discourse.gstreamer.org/t/dynamically-removing-webrtcsrc/5322)

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**Author:** [@FabPoll](https://discourse.gstreamer.org/u/FabPoll)\
**Replies:** 11\
**Last updated:** [September 5, 2025, 4:15pm UTC](https://discourse.gstreamer.org/t/dynamically-removing-webrtcsrc/5322 "2025-09-05T16:15:11Z")

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Hi everyone, I have a question about properly removing elements from a GStreamer pipeline, particularly when using webrtcsrc. I know this topic has been discussed quite often, such as here, but I’m not entirely sure I u…

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## [Pipeline restart on bitrate change](https://discourse.gstreamer.org/t/pipeline-restart-on-bitrate-change/5346)

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**Author:** [@Vavooon](https://discourse.gstreamer.org/u/Vavooon)\
**Replies:** 0\
**Last updated:** [September 4, 2025, 2:01pm UTC](https://discourse.gstreamer.org/t/pipeline-restart-on-bitrate-change/5346 "2025-09-04T14:01:50Z")

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Hello. I have a video decoding pipeline in my Android app: appsrc caps="application/x-rtp, media=video, encoding-name=H264, clock-rate=90000, payload=96" ! rtpjitterbuffer latency=20 ! rtph264depay ! h264parse ! decode…

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## [GStreamer 1 fps recording issue: first 3 seconds show frozen frame](https://discourse.gstreamer.org/t/gstreamer-1-fps-recording-issue-first-3-seconds-show-frozen-frame/5339)

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**Author:** [@RamazanCorduk](https://discourse.gstreamer.org/u/RamazanCorduk)\
**Replies:** 1\
**Last updated:** [September 3, 2025, 6:58am UTC](https://discourse.gstreamer.org/t/gstreamer-1-fps-recording-issue-first-3-seconds-show-frozen-frame/5339 "2025-09-03T06:58:07Z")

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Hello, I’m working on an embedded project on an i.MX8 platform (Yocto Linux). I want to record video at 1 fps to reduce memory usage. My GStreamer pipeline is similar to this: rtspsrc location=rtsp://\<ip\> latency=100 …

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## [Framerate caps seems don't work with gltestsrc](https://discourse.gstreamer.org/t/framerate-caps-seems-dont-work-with-gltestsrc/5337)

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**Author:** [@nuffsaid](https://discourse.gstreamer.org/u/nuffsaid)\
**Replies:** 1\
**Last updated:** [September 2, 2025, 11:20pm UTC](https://discourse.gstreamer.org/t/framerate-caps-seems-dont-work-with-gltestsrc/5337 "2025-09-02T23:20:13Z")

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Hello. Please look at this piece of code… //... //src = gst\_element\_factory\_make("videotestsrc", NULL); src = gst\_element\_factory\_make("gltestsrc", NULL); fpssink = gst\_element\_factory\_make("fpsdisplaysink", NULL); p…

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## [Audio format S20/U20 support in GStreamer Rust bindings?](https://discourse.gstreamer.org/t/audio-format-s20-u20-support-in-gstreamer-rust-bindings/5329)

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**Author:** [@nicholasdezai](https://discourse.gstreamer.org/u/nicholasdezai)\
**Replies:** 2\
**Last updated:** [September 1, 2025, 11:13am UTC](https://discourse.gstreamer.org/t/audio-format-s20-u20-support-in-gstreamer-rust-bindings/5329 "2025-09-01T11:13:42Z")

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Hi all, I am working on an audio backend project (vhost-device-sound) using GStreamer Rust bindings. The device protocol requires 20-bit audio formats (S20/U20) as defined in the virtio-sound specification: /\* analog …

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## [Has I420 been removed in gstreamer nvcodec?](https://discourse.gstreamer.org/t/has-i420-been-removed-in-gstreamer-nvcodec/4983)

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**Author:** [@gfrankliu](https://discourse.gstreamer.org/u/gfrankliu)\
**Replies:** 2\
**Last updated:** [August 17, 2025, 4:34pm UTC](https://discourse.gstreamer.org/t/has-i420-been-removed-in-gstreamer-nvcodec/4983 "2025-08-17T16:34:25Z")

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In Debian 12 with gstreamer 1.22.0, nvcodec has the I420 support: docker run -e NVIDIA\_DRIVER\_CAPABILITIES=compute,video,utility --privileged --gpus "device=GPU-63bc0b91-2265-1b94-831c-15c41fe92675" --rm -it --entrypoin…

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## [Android (Qt 6.9) + GStreamer: how do I build/enable qml6glsink? “Needs a windowing backend” on Android](https://discourse.gstreamer.org/t/android-qt-6-9-gstreamer-how-do-i-build-enable-qml6glsink-needs-a-windowing-backend-on-android/4965)

<div class="topic-metadata">

**Author:** [@raphaelscholle](https://discourse.gstreamer.org/u/raphaelscholle)\
**Replies:** 3\
**Last updated:** [August 17, 2025, 6:45am UTC](https://discourse.gstreamer.org/t/android-qt-6-9-gstreamer-how-do-i-build-enable-qml6glsink-needs-a-windowing-backend-on-android/4965 "2025-08-17T06:45:35Z")

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Goal Get qml6glsink working on Android with Qt 6.9 (QML + GStreamer). Context I’m building GStreamer for Android inside a GitHub Actions runner so the setup is fully reproducible and clean. Workflow is here (exact ste…

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## [Peer to peer stream using gstreamer's webrtcsrc and webrtcsink](https://discourse.gstreamer.org/t/peer-to-peer-stream-using-gstreamers-webrtcsrc-and-webrtcsink/4980)

<div class="topic-metadata">

**Author:** [@d159159](https://discourse.gstreamer.org/u/d159159)\
**Replies:** 0\
**Last updated:** [August 17, 2025, 1:03am UTC](https://discourse.gstreamer.org/t/peer-to-peer-stream-using-gstreamers-webrtcsrc-and-webrtcsink/4980 "2025-08-17T01:03:57Z")

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I want to create peer to peer stream using gstreamer’s webrtcsrc and webrtcsink. I use this command for peer1 : gst-launch-1.0 webrtcsink signaller::uri="ws://51.38.99.21:8443" stun-server="stun://stun.l.google.com:193…

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