mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-09 07:52:36 +00:00
gstaudiobasesink: Break some too long lines
This commit is contained in:
parent
187b106202
commit
2f04733a4b
1 changed files with 14 additions and 13 deletions
|
@ -215,8 +215,8 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * klass)
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
|
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
|
||||||
g_param_spec_int64 ("latency-time", "Latency Time",
|
g_param_spec_int64 ("latency-time", "Latency Time",
|
||||||
"The minimum amount of data to write in each iteration in microseconds",
|
"The minimum amount of data to write in each iteration "
|
||||||
1, G_MAXINT64, DEFAULT_LATENCY_TIME,
|
"in microseconds", 1, G_MAXINT64, DEFAULT_LATENCY_TIME,
|
||||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||||
|
|
||||||
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
|
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
|
||||||
|
@ -555,7 +555,8 @@ gst_audio_base_sink_get_time (GstClock * clock, GstAudioBaseSink * sink)
|
||||||
*
|
*
|
||||||
* 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
|
* 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.
|
* of @sink. If @provide is %FALSE, gst_element_provide_clock() will return
|
||||||
|
* NULL.
|
||||||
*/
|
*/
|
||||||
void
|
void
|
||||||
gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
|
gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
|
||||||
|
@ -1284,8 +1285,8 @@ gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
|
||||||
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
|
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
|
||||||
last_align = sink->priv->last_align;
|
last_align = sink->priv->last_align;
|
||||||
|
|
||||||
/* if we were aligning in the wrong direction or we aligned more than what we
|
/* if we were aligning in the wrong direction or we aligned more than what
|
||||||
* will correct, resync */
|
* we will correct, resync */
|
||||||
if (last_align < 0 || last_align > driftsamples)
|
if (last_align < 0 || last_align > driftsamples)
|
||||||
sink->next_sample = -1;
|
sink->next_sample = -1;
|
||||||
|
|
||||||
|
@ -1312,8 +1313,8 @@ gst_audio_base_sink_skew_slaving (GstAudioBaseSink * sink,
|
||||||
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
|
driftsamples = (sink->ringbuffer->spec.info.rate * mdrift) / GST_SECOND;
|
||||||
last_align = sink->priv->last_align;
|
last_align = sink->priv->last_align;
|
||||||
|
|
||||||
/* if we were aligning in the wrong direction or we aligned more than what we
|
/* if we were aligning in the wrong direction or we aligned more than what
|
||||||
* will correct, resync */
|
* we will correct, resync */
|
||||||
if (last_align > 0 || -last_align > driftsamples)
|
if (last_align > 0 || -last_align > driftsamples)
|
||||||
sink->next_sample = -1;
|
sink->next_sample = -1;
|
||||||
|
|
||||||
|
@ -1780,8 +1781,8 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
|
||||||
}
|
}
|
||||||
|
|
||||||
if (G_UNLIKELY (sync_offset != 0)) {
|
if (G_UNLIKELY (sync_offset != 0)) {
|
||||||
/* compensate for ts-offset and delay we know this will not underflow because we
|
/* compensate for ts-offset and delay. We know this will not underflow
|
||||||
* clipped above. */
|
* because we clipped above. */
|
||||||
GST_DEBUG_OBJECT (sink,
|
GST_DEBUG_OBJECT (sink,
|
||||||
"compensating for sync-offset %" GST_TIME_FORMAT,
|
"compensating for sync-offset %" GST_TIME_FORMAT,
|
||||||
GST_TIME_ARGS (sync_offset));
|
GST_TIME_ARGS (sync_offset));
|
||||||
|
@ -2008,9 +2009,9 @@ sync_latency_failed:
|
||||||
* gst_audio_base_sink_create_ringbuffer:
|
* gst_audio_base_sink_create_ringbuffer:
|
||||||
* @sink: a #GstAudioBaseSink.
|
* @sink: a #GstAudioBaseSink.
|
||||||
*
|
*
|
||||||
* Create and return the #GstAudioRingBuffer for @sink. This function will call the
|
* Create and return the #GstAudioRingBuffer for @sink. This function will
|
||||||
* ::create_ringbuffer vmethod and will set @sink as the parent of the returned
|
* call the ::create_ringbuffer vmethod and will set @sink as the parent of
|
||||||
* buffer (see gst_object_set_parent()).
|
* the returned buffer (see gst_object_set_parent()).
|
||||||
*
|
*
|
||||||
* Returns: (transfer none): The new ringbuffer of @sink.
|
* Returns: (transfer none): The new ringbuffer of @sink.
|
||||||
*/
|
*/
|
||||||
|
@ -2046,7 +2047,7 @@ gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf, guint8 * data,
|
||||||
GST_PAD_STREAM_LOCK (basesink->sinkpad);
|
GST_PAD_STREAM_LOCK (basesink->sinkpad);
|
||||||
|
|
||||||
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
|
/* would be nice to arrange for pad_alloc_buffer to return data -- as it is we
|
||||||
will copy twice, once into data, once into DMA */
|
* will copy twice, once into data, once into DMA */
|
||||||
GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
|
GST_LOG_OBJECT (basesink, "pulling %u bytes offset %" G_GUINT64_FORMAT
|
||||||
" to fill audio buffer", len, basesink->offset);
|
" to fill audio buffer", len, basesink->offset);
|
||||||
ret =
|
ret =
|
||||||
|
|
Loading…
Reference in a new issue