Uncrustify sources
Ran `find src plugins -iname '*.[c|h]' -print0 | xargs -0 uncrustify --no-backup` with some minimal manual intervention.
This commit is contained in:
Evangelos Ribeiro Tzaras
@@ -113,44 +113,44 @@ static uint signals[N_SIGNALS];
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struct _CallsSipMediaPipeline {
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GObject parent;
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GObject parent;
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MediaCodecInfo *codec;
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gboolean debug;
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MediaCodecInfo *codec;
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gboolean debug;
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CallsMediaPipelineState state;
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uint element_map_playing;
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uint element_map_paused;
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uint element_map_stopped;
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gboolean emitted_sending_signal;
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uint element_map_playing;
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uint element_map_paused;
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uint element_map_stopped;
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gboolean emitted_sending_signal;
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/* Connection details */
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char *remote;
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gint rport_rtp;
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gint rport_rtcp;
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char *remote;
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gint rport_rtp;
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gint rport_rtcp;
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GstElement *pipeline;
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GstElement *rtpbin;
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GstElement *pipeline;
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GstElement *rtpbin;
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GstElement *rtp_src;
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GstElement *rtp_sink;
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GstElement *rtcp_sink;
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GstElement *rtcp_src;
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GstElement *rtp_src;
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GstElement *rtp_sink;
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GstElement *rtcp_sink;
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GstElement *rtcp_src;
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GstElement *audio_src;
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GstElement *payloader;
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GstElement *encoder;
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GstElement *audio_src;
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GstElement *payloader;
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GstElement *encoder;
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GstElement *audio_sink;
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GstElement *depayloader;
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GstElement *decoder;
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GstElement *audio_sink;
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GstElement *depayloader;
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GstElement *decoder;
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/* Gstreamer busses */
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GstBus *bus;
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guint bus_watch_id;
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GstBus *bus;
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guint bus_watch_id;
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};
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#if GLIB_CHECK_VERSION(2, 70, 0)
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#if GLIB_CHECK_VERSION (2, 70, 0)
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G_DEFINE_FINAL_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT)
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#else
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G_DEFINE_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT)
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@@ -254,24 +254,24 @@ on_bus_message (GstBus *bus,
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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{
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g_autoptr (GError) error = NULL;
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g_autofree char *msg = NULL;
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{
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g_autoptr (GError) error = NULL;
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g_autofree char *msg = NULL;
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gst_message_parse_error (message, &error, &msg);
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g_warning ("Error on the message bus: %s (%s)", error->message, msg);
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break;
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}
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gst_message_parse_error (message, &error, &msg);
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g_warning ("Error on the message bus: %s (%s)", error->message, msg);
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break;
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}
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case GST_MESSAGE_WARNING:
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{
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g_autoptr (GError) error = NULL;
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g_autofree char *msg = NULL;
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{
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g_autoptr (GError) error = NULL;
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g_autofree char *msg = NULL;
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gst_message_parse_warning (message, &error, &msg);
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g_warning ("Warning on the message bus: %s (%s)", error->message, msg);
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break;
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}
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gst_message_parse_warning (message, &error, &msg);
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g_warning ("Warning on the message bus: %s (%s)", error->message, msg);
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break;
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}
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case GST_MESSAGE_EOS:
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g_debug ("Received end of stream");
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@@ -279,77 +279,77 @@ on_bus_message (GstBus *bus,
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break;
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case GST_MESSAGE_STATE_CHANGED:
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{
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GstState oldstate;
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GstState newstate;
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uint element_id = 0;
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uint unset_element_id;
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{
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GstState oldstate;
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GstState newstate;
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uint element_id = 0;
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uint unset_element_id;
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gst_message_parse_state_changed (message, &oldstate, &newstate, NULL);
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gst_message_parse_state_changed (message, &oldstate, &newstate, NULL);
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g_debug ("Element %s has changed state from %s to %s",
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GST_OBJECT_NAME (message->src),
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gst_element_state_get_name (oldstate),
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gst_element_state_get_name (newstate));
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g_debug ("Element %s has changed state from %s to %s",
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GST_OBJECT_NAME (message->src),
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gst_element_state_get_name (oldstate),
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gst_element_state_get_name (newstate));
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if (message->src == GST_OBJECT (self->pipeline))
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element_id = EL_PIPELINE;
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else if (message->src == GST_OBJECT (self->rtpbin))
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element_id = EL_RTPBIN;
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if (message->src == GST_OBJECT (self->pipeline))
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element_id = EL_PIPELINE;
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else if (message->src == GST_OBJECT (self->rtpbin))
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element_id = EL_RTPBIN;
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else if (message->src == GST_OBJECT (self->rtp_src))
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element_id = EL_RTP_SRC;
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else if (message->src == GST_OBJECT (self->rtp_sink))
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element_id = EL_RTP_SINK;
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else if (message->src == GST_OBJECT (self->rtp_src))
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element_id = EL_RTP_SRC;
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else if (message->src == GST_OBJECT (self->rtp_sink))
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element_id = EL_RTP_SINK;
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else if (message->src == GST_OBJECT (self->rtcp_src))
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element_id = EL_RTCP_SRC;
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else if (message->src == GST_OBJECT (self->rtcp_sink))
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element_id = EL_RTCP_SINK;
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else if (message->src == GST_OBJECT (self->rtcp_src))
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element_id = EL_RTCP_SRC;
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else if (message->src == GST_OBJECT (self->rtcp_sink))
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element_id = EL_RTCP_SINK;
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/* TODO srtp encryption
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else if (message->src == GST_OBJECT (self->srtpenc))
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element_id = EL_SRTP_ENCODER;
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else if (message->src == GST_OBJECT (self->srtpdec))
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element_id = EL_SRTP_DECODER;
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*/
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/* TODO srtp encryption
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else if (message->src == GST_OBJECT (self->srtpenc))
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element_id = EL_SRTP_ENCODER;
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else if (message->src == GST_OBJECT (self->srtpdec))
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element_id = EL_SRTP_DECODER;
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*/
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else if (message->src == GST_OBJECT (self->audio_src))
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element_id = EL_AUDIO_SRC;
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else if (message->src == GST_OBJECT (self->audio_sink))
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element_id = EL_AUDIO_SINK;
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else if (message->src == GST_OBJECT (self->audio_src))
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element_id = EL_AUDIO_SRC;
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else if (message->src == GST_OBJECT (self->audio_sink))
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element_id = EL_AUDIO_SINK;
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else if (message->src == GST_OBJECT (self->payloader))
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element_id = EL_PAYLOADER;
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else if (message->src == GST_OBJECT (self->depayloader))
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element_id = EL_DEPAYLOADER;
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else if (message->src == GST_OBJECT (self->payloader))
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element_id = EL_PAYLOADER;
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else if (message->src == GST_OBJECT (self->depayloader))
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element_id = EL_DEPAYLOADER;
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else if (message->src == GST_OBJECT (self->encoder))
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element_id = EL_ENCODER;
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else if (message->src == GST_OBJECT (self->decoder))
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element_id = EL_DECODER;
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else if (message->src == GST_OBJECT (self->encoder))
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element_id = EL_ENCODER;
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else if (message->src == GST_OBJECT (self->decoder))
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element_id = EL_DECODER;
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unset_element_id = G_MAXUINT ^ element_id;
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unset_element_id = G_MAXUINT ^ element_id;
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if (newstate == GST_STATE_PLAYING) {
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self->element_map_playing |= element_id;
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self->element_map_paused &= unset_element_id;
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self->element_map_stopped &= unset_element_id;
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} else if (newstate == GST_STATE_PAUSED) {
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self->element_map_paused |= element_id;
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self->element_map_playing &= unset_element_id;
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self->element_map_stopped &= unset_element_id;
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} else if (newstate == GST_STATE_NULL) {
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self->element_map_stopped |= element_id;
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self->element_map_playing &= unset_element_id;
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self->element_map_paused &= unset_element_id;
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}
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check_element_maps (self);
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break;
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if (newstate == GST_STATE_PLAYING) {
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self->element_map_playing |= element_id;
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self->element_map_paused &= unset_element_id;
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self->element_map_stopped &= unset_element_id;
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} else if (newstate == GST_STATE_PAUSED) {
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self->element_map_paused |= element_id;
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self->element_map_playing &= unset_element_id;
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self->element_map_stopped &= unset_element_id;
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} else if (newstate == GST_STATE_NULL) {
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self->element_map_stopped |= element_id;
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self->element_map_playing &= unset_element_id;
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self->element_map_paused &= unset_element_id;
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}
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check_element_maps (self);
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break;
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}
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default:
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if (self->debug)
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g_debug ("Got unhandled %s message", GST_MESSAGE_TYPE_NAME (message));
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@@ -584,7 +584,7 @@ pipeline_link_elements (CallsSipMediaPipeline *self,
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gst_object_unref (sinkpad);
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srcpad = gst_element_get_static_pad (self->rtcp_src, "src");
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#if GST_CHECK_VERSION (1, 20 , 0)
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#if GST_CHECK_VERSION (1, 20, 0)
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sinkpad = gst_element_request_pad_simple (self->rtpbin, "recv_rtcp_sink_0");
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#else
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sinkpad = gst_element_get_request_pad (self->rtpbin, "recv_rtcp_sink_0");
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@@ -596,21 +596,21 @@ pipeline_link_elements (CallsSipMediaPipeline *self,
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return FALSE;
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}
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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#if GST_CHECK_VERSION (1, 20, 0)
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srcpad = gst_element_request_pad_simple (self->rtpbin, "send_rtcp_src_0");
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srcpad = gst_element_request_pad_simple (self->rtpbin, "send_rtcp_src_0");
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#else
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srcpad = gst_element_get_request_pad (self->rtpbin, "send_rtcp_src_0");
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srcpad = gst_element_get_request_pad (self->rtpbin, "send_rtcp_src_0");
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#endif
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sinkpad = gst_element_get_static_pad (self->rtcp_sink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpbin to rtcpsink");
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return FALSE;
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}
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sinkpad = gst_element_get_static_pad (self->rtcp_sink, "sink");
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if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK) {
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if (error)
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g_set_error (error, G_IO_ERROR, G_IO_ERROR_FAILED,
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"Failed to link rtpbin to rtcpsink");
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return FALSE;
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}
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/* can only link to depayloader after RTP payload has been verified */
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g_signal_connect (self->rtpbin, "pad-added", G_CALLBACK (on_pad_added), self->depayloader);
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@@ -966,8 +966,7 @@ diagnose_ports_in_use (CallsSipMediaPipeline *self)
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if (same_socket) {
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g_debug ("Diagnosing bidirectional socket...");
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diagnose_used_ports_in_socket (socket_in);
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}
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else {
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} else {
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g_debug ("Diagnosing server socket...");
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diagnose_used_ports_in_socket (socket_in);
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g_debug ("Diagnosing client socket...");
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