diff options
Diffstat (limited to 'gst-libs/gst/audio/gstaudiobasesrc.c')
| -rw-r--r-- | gst-libs/gst/audio/gstaudiobasesrc.c | 31 |
1 files changed, 18 insertions, 13 deletions
diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c index 225184d50..f4890cfa5 100644 --- a/gst-libs/gst/audio/gstaudiobasesrc.c +++ b/gst-libs/gst/audio/gstaudiobasesrc.c @@ -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. */ |