/dports/audio/gstreamer1-plugins-jack/gst-plugins-good-1.16.2/ |
H A D | ChangeLog | 52643 pulsesrc: don't assert in get_time() when called after shutdown 55493 When get_time is called but we are not yet negotiated, return 0 instead of 55495 our get_time is called. 57661 calling get_time with the object lock. 94387 Use the acquired field of the ringbuffer in get_time to know when we are in an 99270 Remove the get_time callback, the GStreamer part will feed us with 106703 pulsesink: handle server disconnect in get_time
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/dports/audio/gstreamer1-plugins-jack/gst-plugins-good-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 191 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-ladspa/gst-plugins-bad-1.16.2/ |
H A D | ChangeLog | 181052 Remove the get_time callback, the GStreamer part will feed us with
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/dports/audio/gstreamer1-plugins-ladspa/gst-plugins-bad-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 450 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-lame/gst-plugins-good-1.16.2/ |
H A D | ChangeLog | 52643 pulsesrc: don't assert in get_time() when called after shutdown 55493 When get_time is called but we are not yet negotiated, return 0 instead of 55495 our get_time is called. 57661 calling get_time with the object lock. 94387 Use the acquired field of the ringbuffer in get_time to know when we are in an 99270 Remove the get_time callback, the GStreamer part will feed us with 106703 pulsesink: handle server disconnect in get_time
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/dports/audio/gstreamer1-plugins-lame/gst-plugins-good-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 191 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-lv2/gst-plugins-bad-1.16.2/ |
H A D | ChangeLog | 181052 Remove the get_time callback, the GStreamer part will feed us with
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/dports/audio/gstreamer1-plugins-lv2/gst-plugins-bad-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 450 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-modplug/gst-plugins-bad-1.16.2/ |
H A D | ChangeLog | 181052 Remove the get_time callback, the GStreamer part will feed us with
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/dports/audio/gstreamer1-plugins-modplug/gst-plugins-bad-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 450 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-mpg123/gst-plugins-good-1.16.2/ |
H A D | ChangeLog | 52643 pulsesrc: don't assert in get_time() when called after shutdown 55493 When get_time is called but we are not yet negotiated, return 0 instead of 55495 our get_time is called. 57661 calling get_time with the object lock. 94387 Use the acquired field of the ringbuffer in get_time to know when we are in an 99270 Remove the get_time callback, the GStreamer part will feed us with 106703 pulsesink: handle server disconnect in get_time
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/dports/audio/gstreamer1-plugins-mpg123/gst-plugins-good-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 191 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-musepack/gst-plugins-bad-1.16.2/ |
H A D | ChangeLog | 181052 Remove the get_time callback, the GStreamer part will feed us with
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/dports/audio/gstreamer1-plugins-musepack/gst-plugins-bad-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 450 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/ |
H A D | ChangeLog | 53172 similar to the way we have used get_time at other places in the code. Acquire it after the 53173 get_time call. This way PA Main loop will be able to post its stream status message by 136144 samples we've written return 0 from the audio sink get_time method.
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/ext/gl/ |
H A D | gstglfilterglass.c | 261 get_time (void) in get_time() function 369 glass_filter->start_time = get_time (); in gst_gl_filter_glass_callback() 372 (glass_filter->timestamp / 1000) - (get_time () - in gst_gl_filter_glass_callback() 387 gint64 time_passed = get_time () - glass_filter->start_time; in gst_gl_filter_glass_callback()
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/dports/audio/gstreamer1-plugins-ogg/gst-plugins-base-1.16.2/gst-libs/gst/rtsp/ |
H A D | gstrtsprange.c | 524 get_time (GstRTSPRangeUnit unit, const GstRTSPTime * t1, in get_time() function 590 *min = get_time (range->unit, &range->min, &range->min2); in gst_rtsp_range_get_times() 592 *max = get_time (range->unit, &range->max, &range->max2); in gst_rtsp_range_get_times() 687 get_time (range->unit, &range->min, &range->min2)); in gst_rtsp_range_convert_units() 689 get_time (range->unit, &range->max, &range->max2)); in gst_rtsp_range_convert_units()
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/dports/audio/gstreamer1-plugins-openmpt/gst-plugins-bad-1.16.2/ |
H A D | ChangeLog | 181052 Remove the get_time callback, the GStreamer part will feed us with
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/dports/audio/gstreamer1-plugins-openmpt/gst-plugins-bad-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 450 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/ |
H A D | ChangeLog | 53172 similar to the way we have used get_time at other places in the code. Acquire it after the 53173 get_time call. This way PA Main loop will be able to post its stream status message by 136144 samples we've written return 0 from the audio sink get_time method.
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/ext/gl/ |
H A D | gstglfilterglass.c | 261 get_time (void) in get_time() function 369 glass_filter->start_time = get_time (); in gst_gl_filter_glass_callback() 372 (glass_filter->timestamp / 1000) - (get_time () - in gst_gl_filter_glass_callback() 387 gint64 time_passed = get_time () - glass_filter->start_time; in gst_gl_filter_glass_callback()
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/dports/audio/gstreamer1-plugins-opus/gst-plugins-base-1.16.2/gst-libs/gst/rtsp/ |
H A D | gstrtsprange.c | 524 get_time (GstRTSPRangeUnit unit, const GstRTSPTime * t1, in get_time() function 590 *min = get_time (range->unit, &range->min, &range->min2); in gst_rtsp_range_get_times() 592 *max = get_time (range->unit, &range->max, &range->max2); in gst_rtsp_range_get_times() 687 get_time (range->unit, &range->min, &range->min2)); in gst_rtsp_range_convert_units() 689 get_time (range->unit, &range->max, &range->max2)); in gst_rtsp_range_convert_units()
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/dports/audio/gstreamer1-plugins-shout2/gst-plugins-good-1.16.2/ |
H A D | ChangeLog | 52643 pulsesrc: don't assert in get_time() when called after shutdown 55493 When get_time is called but we are not yet negotiated, return 0 instead of 55495 our get_time is called. 57661 calling get_time with the object lock. 94387 Use the acquired field of the ringbuffer in get_time to know when we are in an 99270 Remove the get_time callback, the GStreamer part will feed us with 106703 pulsesink: handle server disconnect in get_time
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/dports/audio/gstreamer1-plugins-shout2/gst-plugins-good-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 191 Remove get_time code that is both wrong and unneeded.
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/dports/audio/gstreamer1-plugins-sidplay/gst-plugins-ugly-1.16.2/docs/random/ |
H A D | ChangeLog-0.8 | 382 Remove get_time code that is both wrong and unneeded.
|