diff options
| author | Johan Hedberg <johan.hedberg@nokia.com> | 2008-12-20 21:04:44 +0200 |
|---|---|---|
| committer | Johan Hedberg <johan.hedberg@nokia.com> | 2008-12-20 21:04:44 +0200 |
| commit | b0f85afdba9d51cecc21ac04606a2f759f95df86 (patch) | |
| tree | fe532bf96eba84081a4e6e6f03b30ca91ced4817 /audio/gstavdtpsink.c | |
| parent | 450e5656a8de08051f221d63e026c259151ef274 (diff) | |
Coding style fixes
Diffstat (limited to 'audio/gstavdtpsink.c')
| -rw-r--r-- | audio/gstavdtpsink.c | 19 |
1 files changed, 10 insertions, 9 deletions
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); |