From 5f07c1ed4e58ad52adf53ce58b808a061badd321 Mon Sep 17 00:00:00 2001 From: "Reynaldo H. Verdejo Pinochet" Date: Fri, 20 Dec 2013 19:48:06 -0300 Subject: [PATCH] audiobasesrc: Bunch of cosmetic/grammar fixes --- gst-libs/gst/audio/gstaudiobasesrc.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c index 3e9fb53287..db2f462eca 100644 --- a/gst-libs/gst/audio/gstaudiobasesrc.c +++ b/gst-libs/gst/audio/gstaudiobasesrc.c @@ -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. */ -- 2.34.1