We have collected the most relevant information on Gstreamer Audioconvert Mono. Open the URLs, which are collected below, and you will find all the info you are interested in.
c++ - gstreamer choose one channel and convert to …
https://stackoverflow.com/questions/16472280/gstreamer-choose-one-channel-and-convert-to-mono-deinterleave
ALSA doesn't mind whether you feed mono or stereo data to it, but what if you explicitly wanted a stereo stream/file? Just add audiopanorama . It takes an audio stream and places it somewhere between the left and the right speaker (the panorama parameter defaults to 0 which is the center) and produces a stereo stream:
audioconvert - GStreamer
https://gstreamer.freedesktop.org/documentation/audioconvert/index.html
audioconvert. Audioconvert converts raw audio buffers between various possible formats. It supports integer to float conversion, width/depth conversion, signedness and endianness conversion and channel transformations (ie. upmixing …
audioconvert refuses to convert mono into stereo in ...
https://gstreamer-devel.narkive.com/QzmNNz6Q/audioconvert-refuses-to-convert-mono-into-stereo-in-gstreamer-1-09
*is INVALID caps for mono audio.* channel-mask can be channel-mask=(bitmask)0x0000000000000001 , 0x0000000000000002 , 0x0000000000000004 etc. And regarding the bad pipeline example - GST_DEBUG=3 gst-launch-1.0 audiotestsrc ! avenc_g722 ! avdec_g722 ! audioconvert ! audio/x-raw, channels=2 ! fakesink It did play for me in the latest …
interleave - GStreamer
https://gstreamer.freedesktop.org/documentation/interleave/interleave.html
Merges separate mono inputs into one interleaved stream. This element handles all raw floating point sample formats and all signed integer sample formats. The first caps on one of the sinkpads will set the caps of the output so usually an audioconvert element should be placed before every sinkpad of interleave.
lamemp3enc - GStreamer: open source multimedia …
https://gstreamer.freedesktop.org/documentation/lame/index.html
For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will get resampled to 32 KHz. Use filter caps on the src pad to force a particular sample rate. Example pipelines gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3 Encode a test sine signal to MP3.
gst-plugins-base/gstaudioconvert.c at master · …
https://github.com/GStreamer/gst-plugins-base/blob/master/gst/audioconvert/gstaudioconvert.c
See the GNU. * Library General Public License for more details. * Boston, MA 02110-1301, USA. * Audioconvert converts raw audio buffers between various possible formats. * (ie. upmixing and downmixing), as well as dithering and noise-shaping. * This pipeline converts audio to 8-bit.
deinterleave - gstreamer.freedesktop.org
https://gstreamer.freedesktop.org/documentation/interleave/deinterleave.html
deinterleave. Splits one interleaved multichannel audio stream into many mono audio streams. This element handles all raw audio formats and supports changing the input caps as long as all downstream elements can handle the new caps and the number of channels and the channel positions stay the same.
Audio-channels - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/gstaudiochannels.html
AudioChannelPosition ]): { // javascript wrapper for 'gst_audio_channel_positions_from_mask' } Convert the channels present in channel_mask to a position array (which should have at least channels entries ensured by caller). If channel_mask is set to 0, it is considered as 'not present' for purpose of conversion.
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