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Clocks and synchronization in GStreamer

    https://gstreamer.freedesktop.org/documentation/application-development/advanced/clocks.html#:~:text=This%20is%20used%20in%20audio%20recording%20cases%20where,as%20we%20will%20see%20in%20the%20next%20sections.
    none

Clocks and synchronization in GStreamer

    https://gstreamer.freedesktop.org/documentation/application-development/advanced/clocks.html
    This is used in audio recording cases where you play a previously recorded audio and record new samples, the purpose is to have the new audio perfectly in sync with the previously recorded data. GStreamer uses a GstClock object, buffer timestamps and a SEGMENT event to synchronize streams in a pipeline as we will see in the next sections.

audiorate - GStreamer

    https://gstreamer.freedesktop.org/documentation/audiorate/index.html
    This element takes an incoming stream of timestamped raw audio frames and produces a perfect stream by inserting or dropping samples as needed. This operation may be of use to link to elements that require or otherwise implicitly assume a perfect stream as they do not store timestamps, but derive this by some means (e.g. bitrate for some AVI ...

c++ - Gstreamer recording video with audio - Stack Overflow

    https://stackoverflow.com/questions/55029884/gstreamer-recording-video-with-audio
    What you need is the multiplexer - such GStreamer element that can merge two streams into one. mp4, mkv, avi.. are just a container formats which contains multiple "data streams", which can be audio, video, subtitles (not all formats support this).

About timestamp when recording video using Gstreamer ...

    https://forums.developer.nvidia.com/t/about-timestamp-when-recording-video-using-gstreamer/63693
    Hi, I am new with TX2 and Gstreamer. I am using this kind of Gstreamer command to record raw video data from my own CSI camera. gst-launch-1.0 -e v4l2src device=/dev/video1 num-buffers=450 ! 'video/x-raw, format=UYVY, …

[gstreamer-bugs] [Bug 378397] New: Internal GStreamer ...

    https://gstreamer-bugs.narkive.com/5ehotlpX/bug-378397-new-internal-gstreamer-error-clock-problem-unexpected-discontinuity-in-audio-timestamps
    [gstreamer-bugs] [Bug 378397] New: Internal GStreamer error: clock problem / Unexpected discontinuity in audio timestamps of more than half a second GStreamer (bugzilla.gnome.org) 2006-11-23 07:14:57 UTC

Synchronized audio mixing in GStreamer – coaxion.net ...

    https://coaxion.net/blog/2013/11/synchronized-audio-mixing-in-gstreamer/
    Over the last few weeks I worked on a new GStreamer element: audiomixer.This new element is based on adder, i.e. it mixes multiple audio streams together and produces a single audio stream.It’s already merged into GIT master of the gst-plugins-bad module. The main and important difference to adder is that it actually synchronizes the different audio streams …

Gstreamer example preview, save and stream video · GitHub

    https://gist.github.com/jetsonhacks/1008d24f02e532217cfa
    #!bin/sh # NVIDIA Jetson TK1 # Use Gstreamer to grab H.264 video and audio stream from Logitech c920 webcam # Preview video on screen # Save Video and Audio to a file # Send video as RTSP stream over TCP # IP Address of the this machine hosting the TCP stream IP_ADDRESS= < ENTER IP ADDRESS HERE e.g 10.10.10. 10> # You can list devices: # $ v4l2-ctl --list-devices …

GStreamer Network Video Stream and Save to File - JetsonHacks

    https://www.jetsonhacks.com/2014/10/28/gstreamer-network-video-stream-save-file/
    The first mux, called mp4mux, is a .MP4 file container where we store the video and the audio being collected from the webcam. The video is encoded in H.264, the audio is encoded as AAC. The mp4mux has a place to store video (mp4mux.video_0) and a place where the audio goes (mp4mux.audio_0), and prepares it to go into a file.

AudioEncoder in gstreamer_audio - Rust

    https://gstreamer.pages.freedesktop.org/gstreamer-rs/stable/latest/docs/gstreamer_audio/struct.AudioEncoder.html
    An input timestamp is compared with the expected timestamp as dictated by input sample stream and if the deviation is less than property::AudioEncoder::tolerance, the deviation is discarded. Otherwise, it is considered a discontuinity and subsequent output timestamp is resynced to the new position after performing configured discontinuity ...

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