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sip: media-pipeline: Keep track of pipeline state
This can be used by the media manager to dispose of pipelines which are done.
This commit is contained in:
Evangelos Ribeiro Tzaras
parent
53d6082d64
commit
fe6951c938
3 changed files with 304 additions and 27 deletions
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@ -24,6 +24,7 @@
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#define G_LOG_DOMAIN "CallsSipMediaPipeline"
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#include "calls-media-pipeline-enums.h"
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#include "calls-sip-media-pipeline.h"
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#include "util.h"
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@ -57,6 +58,39 @@
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* Both pipelines are using RTCP.
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*/
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/* The following defines are used to set/reset bitmaps of playing/paused/stop state */
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#define EL_SEND_PIPELINE (1<<0)
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#define EL_SEND_AUDIO_SRC (1<<1)
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#define EL_SEND_RTPBIN (1<<2)
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#define EL_SEND_RTP_SINK (1<<3)
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#define EL_SEND_RTCP_SINK (1<<4)
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#define EL_SEND_RTCP_SRC (1<<5)
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#define EL_SEND_PAYLOADER (1<<6)
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#define EL_SEND_ENCODER (1<<7)
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#define EL_SEND_ALL_RTP EL_SEND_PIPELINE | EL_SEND_AUDIO_SRC | \
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EL_SEND_RTPBIN | EL_SEND_RTP_SINK | EL_SEND_RTCP_SRC | EL_SEND_RTCP_SINK | \
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EL_SEND_PAYLOADER | EL_SEND_ENCODER
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#define EL_SEND_SENDING EL_SEND_AUDIO_SRC | EL_SEND_RTPBIN | EL_SEND_RTP_SINK | \
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EL_SEND_PAYLOADER | EL_SEND_ENCODER
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/* leave some room for more elements to be added later */
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#define EL_RECV_PIPELINE (1<<16)
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#define EL_RECV_AUDIO_SINK (1<<17)
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#define EL_RECV_RTPBIN (1<<18)
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#define EL_RECV_RTP_SRC (1<<19)
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#define EL_RECV_RTCP_SINK (1<<20)
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#define EL_RECV_RTCP_SRC (1<<21)
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#define EL_RECV_DEPAYLOADER (1<<22)
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#define EL_RECV_DECODER (1<<23)
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#define EL_RECV_ALL_RTP EL_RECV_PIPELINE | EL_RECV_AUDIO_SINK | \
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EL_RECV_RTPBIN | EL_RECV_RTP_SRC | EL_RECV_RTCP_SRC | EL_RECV_RTCP_SINK | \
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EL_RECV_DEPAYLOADER | EL_RECV_DECODER
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enum {
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PROP_0,
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PROP_CODEC,
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@ -66,15 +100,30 @@ enum {
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PROP_LPORT_RTCP,
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PROP_RPORT_RTCP,
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PROP_DEBUG,
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PROP_STATE,
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PROP_LAST_PROP,
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};
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enum {
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SENDING_STARTED,
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N_SIGNALS
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};
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static GParamSpec *props[PROP_LAST_PROP];
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static uint signals[N_SIGNALS];
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struct _CallsSipMediaPipeline {
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GObject parent;
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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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/* Connection details */
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char *remote;
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@ -84,8 +133,6 @@ struct _CallsSipMediaPipeline {
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gint rport_rtcp;
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gint lport_rtcp;
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gboolean is_running;
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/* Gstreamer Elements (sending) */
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GstElement *send_pipeline;
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GstElement *audiosrc;
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@ -119,6 +166,56 @@ static void initable_iface_init (GInitableIface *iface);
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G_DEFINE_TYPE_WITH_CODE (CallsSipMediaPipeline, calls_sip_media_pipeline, G_TYPE_OBJECT,
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G_IMPLEMENT_INTERFACE (G_TYPE_INITABLE, initable_iface_init));
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static void
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set_state (CallsSipMediaPipeline *self,
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CallsMediaPipelineState state)
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{
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g_assert (CALLS_SIP_MEDIA_PIPELINE (self));
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if (self->state == state)
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return;
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self->state = state;
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g_object_notify_by_pspec (G_OBJECT (self), props[PROP_STATE]);
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self->emitted_sending_signal = FALSE;
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}
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static void
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check_element_maps (CallsSipMediaPipeline *self)
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{
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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if (self->element_map_playing == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
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g_debug ("All pipeline elements are playing");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAYING);
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return;
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}
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if (self->element_map_paused == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
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g_debug ("All pipeline elements are paused");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PAUSED);
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return;
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}
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if (self->element_map_stopped == (EL_SEND_ALL_RTP | EL_RECV_ALL_RTP)) {
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g_debug ("All pipeline elements are stopped");
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOPPED);
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return;
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}
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if ((self->element_map_playing & (EL_SEND_SENDING)) == (EL_SEND_SENDING) &&
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!self->emitted_sending_signal) {
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g_debug ("Sender pipeline is sending data to %s RTP/RTCP %d/%d",
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self->remote, self->rport_rtp, self->rport_rtcp);
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g_signal_emit (self, signals[SENDING_STARTED], 0);
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self->emitted_sending_signal = TRUE;
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}
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}
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/* rtpbin adds a pad once the payload is verified */
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static void
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on_pad_added (GstElement *rtpbin,
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@ -144,7 +241,7 @@ on_bus_message (GstBus *bus,
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GstMessage *message,
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gpointer data)
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{
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CallsSipMediaPipeline *pipeline = CALLS_SIP_MEDIA_PIPELINE (data);
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CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (data);
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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@ -169,24 +266,79 @@ on_bus_message (GstBus *bus,
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case GST_MESSAGE_EOS:
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g_debug ("Received end of stream");
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calls_sip_media_pipeline_stop (pipeline);
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calls_sip_media_pipeline_stop (self);
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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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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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/* Sender pipeline elements */
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if (message->src == GST_OBJECT (self->send_pipeline))
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element_id = EL_SEND_PIPELINE;
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else if (message->src == GST_OBJECT (self->audiosrc))
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element_id = EL_SEND_AUDIO_SRC;
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else if (message->src == GST_OBJECT (self->send_rtpbin))
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element_id = EL_SEND_RTPBIN;
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else if (message->src == GST_OBJECT (self->rtp_sink))
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element_id = EL_SEND_RTP_SINK;
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else if (message->src == GST_OBJECT (self->rtcp_send_sink))
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element_id = EL_SEND_RTCP_SINK;
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else if (message->src == GST_OBJECT (self->rtcp_send_src))
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element_id = EL_SEND_RTCP_SRC;
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else if (message->src == GST_OBJECT (self->payloader))
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element_id = EL_SEND_PAYLOADER;
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else if (message->src == GST_OBJECT (self->encoder))
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element_id = EL_SEND_ENCODER;
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/* Receiver pipeline elements */
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else if (message->src == GST_OBJECT (self->recv_pipeline))
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element_id = EL_RECV_PIPELINE;
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else if (message->src == GST_OBJECT (self->audiosink))
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element_id = EL_RECV_AUDIO_SINK;
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else if (message->src == GST_OBJECT (self->recv_rtpbin))
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element_id = EL_RECV_RTPBIN;
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else if (message->src == GST_OBJECT (self->rtp_src))
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element_id = EL_RECV_RTP_SRC;
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else if (message->src == GST_OBJECT (self->rtcp_recv_sink))
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element_id = EL_RECV_RTCP_SINK;
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else if (message->src == GST_OBJECT (self->rtcp_recv_src))
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element_id = EL_RECV_RTCP_SRC;
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else if (message->src == GST_OBJECT (self->depayloader))
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element_id = EL_RECV_DEPAYLOADER;
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else if (message->src == GST_OBJECT (self->decoder))
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element_id = EL_RECV_DECODER;
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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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}
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default:
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if (pipeline->debug)
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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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break;
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}
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@ -610,6 +762,10 @@ calls_sip_media_pipeline_get_property (GObject *object,
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g_value_set_boolean (value, self->debug);
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break;
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case PROP_STATE:
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g_value_set_enum (value, calls_sip_media_pipeline_get_state (self));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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@ -733,7 +889,21 @@ calls_sip_media_pipeline_class_init (CallsSipMediaPipelineClass *klass)
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FALSE,
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G_PARAM_READWRITE);
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props[PROP_STATE] = g_param_spec_enum ("state",
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"State",
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"The state of the media pipeline",
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CALLS_TYPE_MEDIA_PIPELINE_STATE,
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CALLS_MEDIA_PIPELINE_STATE_UNKNOWN,
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G_PARAM_READABLE);
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g_object_class_install_properties (object_class, PROP_LAST_PROP, props);
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signals[SENDING_STARTED] =
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g_signal_new ("sending-started",
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G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST,
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0, NULL, NULL, NULL,
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G_TYPE_NONE, 0);
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}
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@ -752,13 +922,20 @@ pipelines_initable_init (GInitable *initable,
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{
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CallsSipMediaPipeline *self = CALLS_SIP_MEDIA_PIPELINE (initable);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_INITIALIZING);
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if (!recv_pipeline_init (self, cancellable, error))
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return FALSE;
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goto err;
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if (!send_pipeline_init (self, cancellable, error))
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return FALSE;
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goto err;
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC);
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return TRUE;
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err:
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
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return FALSE;
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}
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@ -815,17 +992,21 @@ calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
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if (!recv_pipeline_setup_codecs (self, codec, &error)) {
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g_warning ("Error trying to setup codec for receive pipeline: %s",
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error->message);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
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return;
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}
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if (!send_pipeline_setup_codecs (self, codec, &error)) {
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g_warning ("Error trying to setup codec for send pipeline: %s",
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error->message);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_ERROR);
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return;
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}
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self->codec = codec;
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g_object_notify_by_pspec (G_OBJECT (self), props[PROP_CODEC]);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_READY);
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}
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static void
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@ -867,7 +1048,12 @@ diagnose_ports_in_use (CallsSipMediaPipeline *self)
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gboolean same_socket = FALSE;
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g_assert (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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g_assert (self->is_running);
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if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
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self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED) {
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g_warning ("Cannot diagnose ports when pipeline is not active");
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return;
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}
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g_object_get (self->rtp_src, "used-socket", &socket_in, NULL);
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g_object_get (self->rtp_sink, "used-socket", &socket_out, NULL);
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@ -897,13 +1083,12 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
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GSocket *socket;
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g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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if (!self->codec) {
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g_warning ("Codec not set for this pipeline. Cannot start");
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if (self->state != CALLS_MEDIA_PIPELINE_STATE_READY) {
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g_warning ("Cannot start pipeline because it's not ready");
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return;
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}
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g_debug ("Starting media pipeline");
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self->is_running = TRUE;
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/* First start the receiver pipeline so that
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we may reuse the socket in the sender pipeline */
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@ -924,9 +1109,10 @@ calls_sip_media_pipeline_start (CallsSipMediaPipeline *self)
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/* Now start the sender pipeline */
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gst_element_set_state (self->send_pipeline, GST_STATE_PLAYING);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
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if (self->debug)
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diagnose_ports_in_use (self);
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}
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@ -936,11 +1122,12 @@ calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self)
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g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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g_debug ("Stopping media pipeline");
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self->is_running = FALSE;
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/* Stop the pipelines in reverse order (compared to the starting) */
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gst_element_set_state (self->send_pipeline, GST_STATE_NULL);
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gst_element_set_state (self->recv_pipeline, GST_STATE_NULL);
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set_state (self, CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING);
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}
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@ -950,20 +1137,38 @@ calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
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{
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g_return_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self));
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if (self->is_running != pause)
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if (pause &&
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(self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSED ||
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self->state == CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING))
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return;
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g_debug ("%s media pipeline", self->is_running ?
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if (!pause &&
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(self->state == CALLS_MEDIA_PIPELINE_STATE_PLAYING ||
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self->state == CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING))
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return;
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if (self->state != CALLS_MEDIA_PIPELINE_STATE_PLAYING &&
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self->state != CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING &&
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self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSED &&
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self->state != CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING) {
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g_warning ("Cannot pause or unpause pipeline because it's not currently active");
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return;
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}
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g_debug ("%s media pipeline", pause ?
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"Pausing" :
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"Unpausing");
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gst_element_set_state (self->recv_pipeline, self->is_running ?
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gst_element_set_state (self->recv_pipeline, pause ?
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GST_STATE_PAUSED :
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GST_STATE_PLAYING);
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gst_element_set_state (self->send_pipeline, self->is_running ?
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gst_element_set_state (self->send_pipeline, pause ?
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GST_STATE_PAUSED :
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GST_STATE_PLAYING);
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self->is_running = !self->is_running;
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set_state (self, pause ?
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CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING :
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CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING);
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}
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@ -992,4 +1197,38 @@ calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self)
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return port;
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}
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CallsMediaPipelineState
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calls_sip_media_pipeline_get_state (CallsSipMediaPipeline *self)
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{
|
||||
g_return_val_if_fail (CALLS_IS_SIP_MEDIA_PIPELINE (self),
|
||||
CALLS_MEDIA_PIPELINE_STATE_UNKNOWN);
|
||||
|
||||
return self->state;
|
||||
}
|
||||
|
||||
#undef MAKE_ELEMENT
|
||||
|
||||
#undef EL_SEND_PIPELINE
|
||||
#undef EL_SEND_AUDIO_SRC
|
||||
#undef EL_SEND_RTPBIN
|
||||
#undef EL_SEND_RTP_SINK
|
||||
#undef EL_SEND_RTCP_SINK
|
||||
#undef EL_SEND_RTCP_SRC
|
||||
#undef EL_SEND_PAYLOADER
|
||||
#undef EL_SEND_ENCODER
|
||||
|
||||
#undef EL_SEND_ALL_RTP
|
||||
#undef EL_SEND_SENDING
|
||||
|
||||
#undef EL_RECV_PIPELINE
|
||||
#undef EL_RECV_AUDIO_SINK
|
||||
#undef EL_RECV_RTPBIN
|
||||
#undef EL_RECV_RTP_SRC
|
||||
#undef EL_RECV_RTCP_SINK
|
||||
#undef EL_RECV_RTCP_SRC
|
||||
#undef EL_RECV_DEPAYLOADER
|
||||
#undef EL_RECV_DECODER
|
||||
|
||||
#undef EL_RECV_ALL_RTP
|
||||
|
||||
|
|
|
|||
|
|
@ -30,19 +30,49 @@
|
|||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
/**
|
||||
* CallsMediaPipelineState:
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_UNKNOWN: Default state for new pipelines
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_ERROR: Pipeline is in an error state
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_INITIALIZING: Pipeline is initializing
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC: Pipeline was initialized and needs a codec set
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_READY: Pipeline is ready to be set into playing state
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING: Request to start pipeline pending
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_PLAYING: Pipeline is currently playing
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING: Request to pause pipeline pending
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_PAUSED: Pipeline is currently paused
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING: Request to stop pipeline pending
|
||||
* @CALLS_MEDIA_PIPELINE_STATE_STOPPED: Pipeline has stopped playing (f.e. received BYE packet)
|
||||
*/
|
||||
typedef enum {
|
||||
CALLS_MEDIA_PIPELINE_STATE_UNKNOWN = 0,
|
||||
CALLS_MEDIA_PIPELINE_STATE_ERROR,
|
||||
CALLS_MEDIA_PIPELINE_STATE_INITIALIZING,
|
||||
CALLS_MEDIA_PIPELINE_STATE_NEED_CODEC,
|
||||
CALLS_MEDIA_PIPELINE_STATE_READY,
|
||||
CALLS_MEDIA_PIPELINE_STATE_PLAY_PENDING,
|
||||
CALLS_MEDIA_PIPELINE_STATE_PLAYING,
|
||||
CALLS_MEDIA_PIPELINE_STATE_PAUSE_PENDING,
|
||||
CALLS_MEDIA_PIPELINE_STATE_PAUSED,
|
||||
CALLS_MEDIA_PIPELINE_STATE_STOP_PENDING,
|
||||
CALLS_MEDIA_PIPELINE_STATE_STOPPED
|
||||
} CallsMediaPipelineState;
|
||||
|
||||
|
||||
#define CALLS_TYPE_SIP_MEDIA_PIPELINE (calls_sip_media_pipeline_get_type ())
|
||||
|
||||
G_DECLARE_FINAL_TYPE (CallsSipMediaPipeline, calls_sip_media_pipeline, CALLS, SIP_MEDIA_PIPELINE, GObject)
|
||||
|
||||
|
||||
CallsSipMediaPipeline* calls_sip_media_pipeline_new (MediaCodecInfo *codec);
|
||||
void calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
|
||||
MediaCodecInfo *info);
|
||||
void calls_sip_media_pipeline_start (CallsSipMediaPipeline *self);
|
||||
void calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self);
|
||||
void calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
|
||||
gboolean pause);
|
||||
int calls_sip_media_pipeline_get_rtp_port (CallsSipMediaPipeline *self);
|
||||
int calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self);
|
||||
CallsSipMediaPipeline* calls_sip_media_pipeline_new (MediaCodecInfo *codec);
|
||||
void calls_sip_media_pipeline_set_codec (CallsSipMediaPipeline *self,
|
||||
MediaCodecInfo *info);
|
||||
void calls_sip_media_pipeline_start (CallsSipMediaPipeline *self);
|
||||
void calls_sip_media_pipeline_stop (CallsSipMediaPipeline *self);
|
||||
void calls_sip_media_pipeline_pause (CallsSipMediaPipeline *self,
|
||||
gboolean pause);
|
||||
int calls_sip_media_pipeline_get_rtp_port (CallsSipMediaPipeline *self);
|
||||
int calls_sip_media_pipeline_get_rtcp_port (CallsSipMediaPipeline *self);
|
||||
CallsMediaPipelineState calls_sip_media_pipeline_get_state (CallsSipMediaPipeline *self);
|
||||
|
||||
G_END_DECLS
|
||||
|
|
|
|||
|
|
@ -55,6 +55,14 @@ sip_sources = files(
|
|||
]
|
||||
)
|
||||
|
||||
pipeline_enum_headers = [
|
||||
'calls-sip-media-pipeline.h',
|
||||
]
|
||||
|
||||
pipeline_enums = gnome.mkenums_simple('calls-media-pipeline-enums',
|
||||
sources: pipeline_enum_headers)
|
||||
sip_sources += pipeline_enums
|
||||
|
||||
sip_enum_headers = [
|
||||
'calls-sip-util.h',
|
||||
]
|
||||
|
|
|
|||
Loading…
Reference in a new issue