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audio buffer - GStreamer
https://gstreamer.freedesktop.org/documentation/audio/audio-buffer.html
The buffer to clip. segment – Segment in GST_FORMAT_TIME or GST_FORMAT_DEFAULT to which the buffer should be clipped. rate – sample rate. bpf – size of one audio frame in bytes. This is the size of one sample * number of channels. Returns ( [transfer: full]) –
gstaudio - freedesktop.org
https://gstreamer.freedesktop.org/data/doc/gstreamer/1.0/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudio.html
gst_audio_buffer_clip GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer, GstSegment *segment, gint rate, gint bpf); Clip the buffer to the given GstSegment. After calling this function the caller does not own a reference to buffer anymore. buffer : The buffer to clip. [transfer full] segment : Segment in ...
Gst.Audio – gstreamer-audio-1.0 - Valadoc.org
https://valadoc.org/gstreamer-audio-1.0/Gst.Audio.html
This function is set with gst_audio_ring_buffer_set_callback and is called to fill the memory at data with data.length bytes of samples. Functions: public Buffer audio_buffer_clip (owned Buffer buffer, Segment segment, int rate, int bpf) public bool audio_buffer_map (out Buffer buffer, Info info, Buffer gstbuffer, MapFlags flags)
gst-plugins-base/audio.c at master · GStreamer/gst …
https://github.com/GStreamer/gst-plugins-base/blob/master/gst-libs/gst/audio/audio.c
# endif /* GST_DISABLE_GST_DEBUG */ /* * * gst_audio_buffer_clip: * @buffer: (transfer full): The buffer to clip. * @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which * the buffer should be clipped. * @rate: sample rate. * @bpf: size of one audio frame in bytes. This is the size of one sample *
GstAudio: GStreamer Base Plugins 1.0 Library Reference Manual
https://www.manpagez.com/html/gst-plugins-base-libs-1.0/gst-plugins-base-libs-1.0-1.14.0/gst-plugins-base-libs-GstAudio.php
gst_audio_buffer_clip () GstBuffer * gst_audio_buffer_clip (GstBuffer *buffer, const GstSegment *segment, gint rate, gint bpf); Clip the buffer to the given GstSegment. After calling this function the caller does not own a reference to buffer anymore.
gst-plugins-base/gstaudiofilter.c at master · GStreamer ...
https://github.com/GStreamer/gst-plugins-base/blob/master/gst-libs/gst/audio/gstaudiofilter.c
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