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audiobasesrc: Break some too long lines

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This commit is contained in:
Reynaldo H. Verdejo Pinochet 2013-12-20 18:53:13 -03:00
parent 6b17d86692
commit d1b3454299
1 changed files with 18 additions and 13 deletions
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31
gst-libs/gst/audio/gstaudiobasesrc.c
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@ -176,9 +176,9 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"The minimum amount of data to read in each iteration in microseconds, "
"this is the minimum latency that the source reports", 1,
G_MAXINT64, DEFAULT_LATENCY_TIME,
"The minimum amount of data to read in each iteration in "
"microseconds, this is the minimum latency that the source reports",
1, G_MAXINT64, DEFAULT_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
@ -895,7 +895,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
* last written segment in the ringbuffer
* (one segment written = segment 0) */
last_written_segment = segments_written - ringbuffer->segbase - 1;
/* samples per segment */
@ -910,7 +911,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
/* the running_time converted to a sample
* (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, rate, GST_SECOND);
@ -920,7 +922,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
/* the segment currently read from the ringbuffer */
last_read_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime (last written to) */
/* the skew we have between running_time and the ringbuffertime
* (last written to) */
segment_skew = running_time_segment - last_written_segment;
GST_DEBUG_OBJECT (bsrc,
@ -983,9 +986,10 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
/* We are slaved to another clock, take running time of the pipeline
* clock and timestamp against it. Somebody else in the pipeline should
* figure out the clock drift. We keep the duration we calculated
* above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
@ -1011,7 +1015,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
/* the read method returned a timestamp so we use this instead */
timestamp = rb_timestamp;
} else {
/* to get the timestamp against the clock we also need to add our offset */
/* to get the timestamp against the clock we also need to add our
* offset */
timestamp = gst_audio_clock_adjust (clock, timestamp);
}
@ -1085,9 +1090,9 @@ got_error:
* gst_audio_base_src_create_ringbuffer:
* @src: a #GstAudioBaseSrc.
*
* Create and return the #GstAudioRingBuffer for @src. This function will call the
* ::create_ringbuffer vmethod and will set @src as the parent of the returned
* buffer (see gst_object_set_parent()).
* Create and return the #GstAudioRingBuffer for @src. This function will call
* the ::create_ringbuffer vmethod and will set @src as the parent of the
* returned buffer (see gst_object_set_parent()).
*
* Returns: (transfer none): The new ringbuffer of @src.
*/