Gst.RTP.BaseAudioPayload.set_samplebits_options – gstreamer-rtp-1.0 Reference Manual
Packages
gstreamer-rtp-1.0
Gst
RTP
BaseAudioPayload
buffer_list
BaseAudioPayload
flush
get_adapter
push
set_frame_based
set_frame_options
set_sample_based
set_sample_options
set_samplebits_options
base_ts
frame_duration
frame_size
sample_size
set_samplebits_options
Description:
public
void
set_samplebits_options
(
int
sample_size)
Sets the options for sample based audio codecs.
Parameters:
this
a pointer to the element.
sample_size
Size per sample in bits.