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Original commit message from CVS: * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init): This clock can be slaved to a master clock now. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_set_clock), (gst_base_audio_sink_set_property), (gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst-libs/gst/audio/gstbaseaudiosink.h: Handle slaving the internal clock to the clock selected in the pipeline. Add property to make the basesink not provide a clock. * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_wait): * gst-libs/gst/rtp/gstbasertpdepayload.h: We can use the clock in GstElement, no need to store it ourselves. |
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gstbasertpdepayload.c | ||
gstbasertpdepayload.h | ||
gstbasertppayload.c | ||
gstbasertppayload.h | ||
gstrtpbuffer.c | ||
gstrtpbuffer.h | ||
Makefile.am | ||
README |
The RTP libraries --------------------- GstRTPBuffer: A GstBuffer subclass that can has extra RTP information such as timestamps and marks. It is used for communications between the RTPSession element and the RTP payloaders/depayloaders.