From 2f04733a4b8baa64bf27c5fe1e639128596acf68 Mon Sep 17 00:00:00 2001 From: "Reynaldo H. Verdejo Pinochet" Date: Fri, 20 Dec 2013 01:03:14 -0300 Subject: [PATCH] gstaudiobasesink: Break some too long lines --- gst-libs/gst/audio/gstaudiobasesink.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/gst-libs/gst/audio/gstaudiobasesink.c b/gst-libs/gst/audio/gstaudiobasesink.c index 368e4f53ce..9beb1514f2 100644 --- a/gst-libs/gst/audio/gstaudiobasesink.c +++ b/gst-libs/gst/audio/gstaudiobasesink.c @@ -215,8 +215,8 @@ gst_audio_base_sink_class_init (GstAudioBaseSinkClass * 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 write in each iteration in microseconds", - 1, G_MAXINT64, DEFAULT_LATENCY_TIME, + "The minimum amount of data to write in each iteration " + "in microseconds", 1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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, * 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 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; last_align = sink->priv->last_align; - /* if we were aligning in the wrong direction or we aligned more than what we - * will correct, resync */ + /* if we were aligning in the wrong direction or we aligned more than what + * we will correct, resync */ if (last_align < 0 || last_align > driftsamples) 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; last_align = sink->priv->last_align; - /* if we were aligning in the wrong direction or we aligned more than what we - * will correct, resync */ + /* if we were aligning in the wrong direction or we aligned more than what + * we will correct, resync */ if (last_align > 0 || -last_align > driftsamples) sink->next_sample = -1; @@ -1780,8 +1781,8 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf) } if (G_UNLIKELY (sync_offset != 0)) { - /* compensate for ts-offset and delay we know this will not underflow because we - * clipped above. */ + /* compensate for ts-offset and delay. We know this will not underflow + * because we clipped above. */ GST_DEBUG_OBJECT (sink, "compensating for sync-offset %" GST_TIME_FORMAT, GST_TIME_ARGS (sync_offset)); @@ -2008,9 +2009,9 @@ sync_latency_failed: * gst_audio_base_sink_create_ringbuffer: * @sink: a #GstAudioBaseSink. * - * Create and return the #GstAudioRingBuffer for @sink. This function will call the - * ::create_ringbuffer vmethod and will set @sink as the parent of the returned - * buffer (see gst_object_set_parent()). + * Create and return the #GstAudioRingBuffer for @sink. This function will + * call the ::create_ringbuffer vmethod and will set @sink as the parent of + * the returned buffer (see gst_object_set_parent()). * * 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); /* 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 " to fill audio buffer", len, basesink->offset); ret =