Coding style fixes

4 files changed, 14 insertions, 14 deletions

diff --git a/audio/gsta2dpsink.c b/audio/gsta2dpsink.c
index d25cbded..00180cc8 100644
--- a/audio/gsta2dpsink.c
+++ b/audio/gsta2dpsink.c
@@ -97,8 +97,8 @@ static GstState gst_a2dp_sink_get_state(GstA2dpSink *self)
  * Helper function to create elements, add to the bin and link it
  * to another element.
  */
-static GstElement* gst_a2dp_sink_init_element(GstA2dpSink *self,
-			const gchar* elementname, const gchar* name,
+static GstElement *gst_a2dp_sink_init_element(GstA2dpSink *self,
+			const gchar *elementname, const gchar *name,
 			GstElement *link_to)
 {
 	GstElement *element;
@@ -693,7 +693,7 @@ static void gst_a2dp_sink_init(GstA2dpSink *self,
 
 }
 
-gboolean gst_a2dp_sink_plugin_init (GstPlugin * plugin)
+gboolean gst_a2dp_sink_plugin_init (GstPlugin *plugin)
 {
 	return gst_element_register (plugin, "a2dpsink",
 			GST_RANK_PRIMARY, GST_TYPE_A2DP_SINK);
diff --git a/audio/gstavdtpsink.c b/audio/gstavdtpsink.c
index 45a53b5d..311b4425 100644
--- a/audio/gstavdtpsink.c
+++ b/audio/gstavdtpsink.c
@@ -819,7 +819,7 @@ static gint gst_avdtp_sink_get_channel_mode(const gchar *mode)
 }
 
 static void gst_avdtp_sink_tag(const GstTagList *taglist,
-			const gchar* tag, gpointer user_data)
+			const gchar *tag, gpointer user_data)
 {
 	gboolean crc;
 	gchar *channel_mode = NULL;
@@ -962,7 +962,7 @@ static gboolean gst_avdtp_sink_init_mp3_pkt_conf(
 {
 	const GValue *value = NULL;
 	gint rate, layer;
-	const gchar* name;
+	const gchar *name;
 	GstStructure *structure = gst_caps_get_structure(caps, 0);
 
 	name = gst_structure_get_name(structure);
@@ -1119,8 +1119,9 @@ static GstFlowReturn gst_avdtp_sink_render(GstBaseSink *basesink,
 	gsize ret;
 	GIOError err;
 
-	err = g_io_channel_write(self->stream, (gchar*)GST_BUFFER_DATA(buffer),
-			 (gsize)(GST_BUFFER_SIZE(buffer)), &ret);
+	err = g_io_channel_write(self->stream,
+				(gchar *) GST_BUFFER_DATA(buffer),
+				(gsize) (GST_BUFFER_SIZE(buffer)), &ret);
 
 	if (err != G_IO_ERROR_NONE) {
 		GST_ERROR_OBJECT(self, "Error while writting to socket: %d %s",
@@ -1142,7 +1143,7 @@ static gboolean gst_avdtp_sink_unlock(GstBaseSink *basesink)
 }
 
 static GstFlowReturn gst_avdtp_sink_buffer_alloc(GstBaseSink *basesink,
-				guint64 offset, guint size, GstCaps* caps,
+				guint64 offset, guint size, GstCaps *caps,
 				GstBuffer **buf)
 {
 	GstAvdtpSink *self = GST_AVDTP_SINK(basesink);
@@ -1235,7 +1236,7 @@ static GIOError gst_avdtp_sink_audioservice_send(
 
 	length = msg->length ? msg->length : BT_SUGGESTED_BUFFER_SIZE;
 
-	error = g_io_channel_write(self->server, (const gchar*) msg, length,
+	error = g_io_channel_write(self->server, (const gchar *) msg, length,
 								&written);
 	if (error != G_IO_ERROR_NONE)
 		GST_ERROR_OBJECT(self, "Error sending data to audio service:"
@@ -1260,7 +1261,7 @@ static GIOError gst_avdtp_sink_audioservice_recv(
 
 	length = inmsg->length ? inmsg->length : BT_SUGGESTED_BUFFER_SIZE;
 
-	status = g_io_channel_read(self->server, (gchar*) inmsg, length,
+	status = g_io_channel_read(self->server, (gchar *) inmsg, length,
 								&bytes_read);
 	if (status != G_IO_ERROR_NONE) {
 		GST_ERROR_OBJECT(self, "Error receiving data from "
@@ -1315,7 +1316,7 @@ static GIOError gst_avdtp_sink_audioservice_expect(
 	return status;
 }
 
-gboolean gst_avdtp_sink_plugin_init (GstPlugin * plugin)
+gboolean gst_avdtp_sink_plugin_init (GstPlugin *plugin)
 {
 	return gst_element_register (plugin, "avdtpsink",
 			GST_RANK_NONE, GST_TYPE_AVDTP_SINK);
@@ -1354,7 +1355,7 @@ guint gst_avdtp_sink_get_link_mtu(GstAvdtpSink *sink)
 	return sink->data->link_mtu;
 }
 
-void gst_avdtp_sink_set_device(GstAvdtpSink *self, const gchar* dev)
+void gst_avdtp_sink_set_device(GstAvdtpSink *self, const gchar *dev)
 {
 	if (self->device != NULL)
 		g_free(self->device);
diff --git a/audio/ipc.h b/audio/ipc.h
index 0e985c3a..5e3d3985 100644
--- a/audio/ipc.h
+++ b/audio/ipc.h
@@ -198,7 +198,6 @@ typedef struct {
 	uint16_t sampling_rate;
 } __attribute__ ((packed)) pcm_capabilities_t;
 
-
 struct bt_get_capabilities_rsp {
 	bt_audio_msg_header_t	h;
 	uint8_t			data[0];	/* First codec_capabilities_t */
diff --git a/audio/unix.c b/audio/unix.c
index dcfe932d..4378bb24 100644
--- a/audio/unix.c
+++ b/audio/unix.c
@@ -281,7 +281,7 @@ static void headset_setup_complete(struct audio_device *dev, void *user_data)
 	rsp->h.name = BT_SET_CONFIGURATION;
 	rsp->h.length = sizeof(*rsp);
 
-	rsp->transport  = BT_CAPABILITIES_TRANSPORT_SCO;
+	rsp->transport = BT_CAPABILITIES_TRANSPORT_SCO;
 	rsp->access_mode = client->access_mode;
 	rsp->link_mtu = 48;