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baseaudiosink: correct for clock reset

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When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.

Fixes #594136
This commit is contained in:
Wim Taymans 2009-09-09 16:19:32 +02:00
parent 47550f6984
commit fe47c6c4d5
1 changed files with 21 additions and 8 deletions
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29
gst-libs/gst/audio/gstbaseaudiosink.c
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@ -2,7 +2,7 @@
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosink.c:
* gstbaseaudiosink.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -67,7 +67,7 @@ enum
/* we tollerate half a second diff before we start resyncing. This
* should be enough to compensate for various rounding errors in the timestamp
* and sample offset position.
* and sample offset position.
* This is an emergency resync fallback since buffers marked as DISCONT will
* always lock to the correct timestamp immediatly and buffers not marked as
* DISCONT are contiguous by definition.
@ -474,7 +474,7 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
* @sink: a #GstBaseAudioSink
* @provide: new state
*
* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
* Controls whether @sink will provide a clock or not. If @provide is %TRUE,
* gst_element_provide_clock() will return a clock that reflects the datarate
* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
*
@ -521,7 +521,7 @@ gst_base_audio_sink_get_provide_clock (GstBaseAudioSink * sink)
* @sink: a #GstBaseAudioSink
* @method: the new slave method
*
* Controls how clock slaving will be performed in @sink.
* Controls how clock slaving will be performed in @sink.
*
* Since: 0.10.16
*/
@ -658,7 +658,7 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
gst_ring_buffer_activate (sink->ringbuffer, TRUE);
}
/* calculate actual latency and buffer times.
/* calculate actual latency and buffer times.
* FIXME: In 0.11, store the latency_time internally in ns */
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
@ -1115,7 +1115,7 @@ gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %"
GST_TIME_FORMAT, GST_TIME_ARGS (time));
/* wait for the clock, this can be interrupted because we got shut down or
/* wait for the clock, this can be interrupted because we got shut down or
* we PAUSED. */
status = gst_base_sink_wait_clock (bsink, time, &jitter);
@ -1220,6 +1220,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
gboolean sync, slaved, align_next;
GstFlowReturn ret;
GstSegment clip_seg;
gint64 time_offset;
sink = GST_BASE_AUDIO_SINK (bsink);
@ -1310,8 +1311,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
}
/* samples should be rendered based on their timestamp. All samples
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* boundaries */
if (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop, &ctime,
&cstop))
@ -1401,6 +1402,18 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
&render_start, &render_stop);
}
/* bring to position in the ringbuffer */
time_offset =
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset;
if (render_start > time_offset)
render_start -= time_offset;
else
render_start = 0;
if (render_stop > time_offset)
render_stop -= time_offset;
else
render_stop = 0;
/* and bring the time to the rate corrected offset in the buffer */
render_start = gst_util_uint64_scale_int (render_start,
ringbuf->spec.rate, GST_SECOND);