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audiobasesrc: Bunch of cosmetic/grammar fixes

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This commit is contained in:
Reynaldo H. Verdejo Pinochet 2013-12-20 19:48:06 -03:00
parent 0a6d6e1fff
commit 5f07c1ed4e
1 changed files with 13 additions and 13 deletions
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26
gst-libs/gst/audio/gstaudiobasesrc.c
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@ -169,7 +169,7 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
/* FIXME: 2.0, handle BUFFER_TIME and LATENCY in nanoseconds */
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in microseconds, this is the maximum amount "
"Size of audio buffer in microseconds. This is the maximum amount "
"of data that is buffered in the device and the maximum latency that "
"the source reports", 1, G_MAXINT64, DEFAULT_BUFFER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -177,7 +177,7 @@ 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",
"microseconds. This is the minimum latency that the source reports",
1, G_MAXINT64, DEFAULT_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -210,7 +210,7 @@ gst_audio_base_src_class_init (GstAudioBaseSrcClass * klass)
g_object_class_install_property (gobject_class, PROP_SLAVE_METHOD,
g_param_spec_enum ("slave-method", "Slave Method",
"Algorithm to use to match the rate of the masterclock",
"Algorithm used to match the rate of the masterclock",
GST_TYPE_AUDIO_BASE_SRC_SLAVE_METHOD, DEFAULT_SLAVE_METHOD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
@ -595,7 +595,7 @@ static void
gst_audio_base_src_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* no need to sync to a clock here, we schedule the samples based
/* No need to sync to a clock here. We schedule the samples based
* on our own clock for the moment. */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
@ -647,7 +647,7 @@ gst_audio_base_src_query (GstBaseSrc * bsrc, GstQuery * query)
}
case GST_QUERY_SCHEDULING:
{
/* We allow limited pull base operation. Basically pulling can be
/* We allow limited pull base operation. Basically, pulling can be
* done on any number of bytes as long as the offset is -1 or
* sequentially increasing. */
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEQUENTIAL, 1, -1,
@ -701,7 +701,7 @@ gst_audio_base_src_event (GstBaseSrc * bsrc, GstEvent * event)
return res;
}
/* get the next offset in the ringbuffer for reading samples.
/* Get the next offset in the ringbuffer for reading samples.
* If the next sample is too far away, this function will position itself to the
* next most recent sample, creating discontinuity */
static guint64
@ -728,9 +728,9 @@ gst_audio_base_src_get_offset (GstAudioBaseSrc * src)
* the sample should be read from. */
readseg = sample / sps;
/* see how far away it is from the read segment, normally segdone (where new
* data is written in the ringbuffer) is bigger than readseg (where we are
* reading). */
/* See how far away it is from the read segment. Normally, segdone (where
* new data is written in the ringbuffer) is bigger than readseg
* (where we are reading). */
diff = segdone - readseg;
if (diff >= segtotal) {
GST_DEBUG_OBJECT (src, "dropped, align to segment %d", segdone);
@ -796,7 +796,7 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
/* Calculate the sequentially-next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_audio_base_src_get_offset (src);
}
@ -875,8 +875,8 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
/* Not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produce more or less samples based
* on the clock drift */
case GST_AUDIO_BASE_SRC_SLAVE_SKEW:
{
@ -986,7 +986,7 @@ 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
/* 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. */