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audioconvert - GStreamer
https://gstreamer.freedesktop.org/documentation/audioconvert/index.html
gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE This pipeline converts audio to 8-bit. The level element shows that the output levels still match the one for a sine wave.
GstAudioConverter - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/gstaudioconverter.html
gst_audio_converter_new GstAudioConverter * gst_audio_converter_new (GstAudioConverterFlags flags, GstAudioInfo * in_info, GstAudioInfo * out_info, GstStructure * config). Create a new GstAudioConverter that is able to convert between in and out audio formats.. config contains extra configuration options, see GST_AUDIO_CONVERTER_OPT_* …
gst-plugins-base/gstaudioconvert.c at master · …
https://github.com/GStreamer/gst-plugins-base/blob/master/gst/audioconvert/gstaudioconvert.c
* gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE * This pipeline converts audio to 8-bit. The level element shows that
gst-plugins-good/audioinvert.c at master · GStreamer/gst ...
https://github.com/GStreamer/gst-plugins-good/blob/master/gst/audiofx/audioinvert.c
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert degree=0.4 ! alsasink * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioinvert degree=0.4 ! audioconvert ! alsasink
audioconvert: gst_buffer_resize_range: assertion 'bufmax ...
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/912
audioconvert: gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed with pipewiresink pool buffer This happens in main and with 1.18.5 when playing a wma file using the libav decoder, and using pipewiresink as sink (doesn't happen with pulsesink/alsasink for me).
c - GStreamer audio streaming on Windows - Stack …
https://stackoverflow.com/questions/4314633/gstreamer-audio-streaming-on-windows
gst-launch -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink host=224.0.0.7 auto-multicast=true port=4444 This is the receiver pipeline: gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444 caps="application/x-rtp, media=(string)audio, channels=(int)1, clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay ...
Convert gstreamer command to C code - Stack Overflow
https://stackoverflow.com/questions/41698656/convert-gstreamer-command-to-c-code
I have a basic gstreamer command for playing audio stream received from network: gst-launch-1.0 tcpserversrc host=127.0.0.1 port=5000 ! decodebin ! audioconvert ! alsasink I tried to convert it t...
audioconvert
https://www.freedesktop.org/software/gstreamer-sdk/data/docs/2012.5/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-audioconvert.html
This pipeline converts audio to 8-bit. The level element shows that the output levels still match the one for a sine wave. 1. gst-launch -v -m audiotestsrc ! audioconvert ! vorbisenc ! fakesink silent= TRUE. The vorbis encoder takes float audio data instead of the integer data generated by audiotestsrc.
Example GStreamer Pipelines - IGEP - ISEE Wiki
https://labs.isee.biz/index.php?title=Example_GStreamer_Pipelines
gst-launch --gst-debug=audiotestsrc:4 audiotestsrc ! alsasink Adjust the element name and debug level until you get the data you are looking for. To enable debug output for a more than one element: gst-launch --gst-debug=audio*:3 audiotestsrc ! audioconvert ! alsasink or for all elements gst-launch --gst-debug=*:3 alsasrc ! alsasink
GStreamer / [gst-embedded] [PATCH] audioconvert: add …
https://sourceforge.net/p/gstreamer/mailman/message/23290074/
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