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multiparty sendrecv: Add a queue before the audio sink

Browse source 
Missed this, fixes the bug where removing a peer causes the pipeline to
get stuck. However, when peers leave, there is still a chance that the
pipeline will get stuck.
This commit is contained in:
Nirbheek Chauhan 2017-10-30 13:24:21 +05:30
parent 9b1a0e5389
commit 0c5e799952
1 changed files with 64 additions and 15 deletions
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79
webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
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@ -220,7 +220,7 @@ send_room_peer_sdp (GstWebRTCSessionDescription * desc, const gchar * peer_id)
JsonObject *msg, *sdp;
gchar *text, *sdptype, *sdptext;
g_assert_cmpint (app_state, <, ROOM_CALL_OFFERING);
g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING);
if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER)
sdptype = "offer";
@ -284,24 +284,71 @@ on_negotiation_needed (GstElement * webrtc, const gchar * peer_id)
}
static void
add_webrtcbin_to_pipeline (const gchar * peer_id, gboolean offer)
remove_peer_from_pipeline (const gchar * peer_id)
{
gchar *qname;
GstPad *srcpad, *sinkpad;
GstElement *webrtc, *q, *tee;
webrtc = gst_bin_get_by_name (GST_BIN (pipeline), peer_id);
if (!webrtc)
return;
gst_bin_remove (GST_BIN (pipeline), webrtc);
gst_object_unref (webrtc);
qname = g_strdup_printf ("queue-%s", peer_id);
q = gst_bin_get_by_name (GST_BIN (pipeline), qname);
g_free (qname);
sinkpad = gst_element_get_static_pad (q, "sink");
g_assert_nonnull (sinkpad);
srcpad = gst_pad_get_peer (sinkpad);
g_assert_nonnull (srcpad);
gst_object_unref (sinkpad);
gst_bin_remove (GST_BIN (pipeline), q);
gst_object_unref (q);
tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee");
g_assert_nonnull (tee);
gst_element_release_request_pad (tee, srcpad);
gst_object_unref (srcpad);
gst_object_unref (tee);
}
static void
add_peer_to_pipeline (const gchar * peer_id, gboolean offer)
{
int ret;
GstElement *tee, *webrtc;
gchar *tmp;
GstElement *tee, *webrtc, *q;
GstPad *srcpad, *sinkpad;
GError *error = NULL;
tmp = g_strdup_printf ("queue-%s", peer_id);
q = gst_element_factory_make ("queue", tmp);
g_free (tmp);
webrtc = gst_element_factory_make ("webrtcbin", peer_id);
g_assert_nonnull (webrtc);
gst_bin_add_many (GST_BIN (pipeline), q, webrtc, NULL);
srcpad = gst_element_get_static_pad (q, "src");
g_assert_nonnull (srcpad);
sinkpad = gst_element_get_request_pad (webrtc, "sink_%u");
g_assert_nonnull (sinkpad);
ret = gst_pad_link (srcpad, sinkpad);
g_assert_cmpint (ret, ==, GST_PAD_LINK_OK);
gst_object_unref (srcpad);
gst_object_unref (sinkpad);
tee = gst_bin_get_by_name (GST_BIN (pipeline), "audiotee");
g_assert_nonnull (tee);
srcpad = gst_element_get_request_pad (tee, "src_%u");
g_assert_nonnull (srcpad);
gst_object_unref (tee);
sinkpad = gst_element_get_request_pad (webrtc, "sink_%u");
/* Add the bin to the pipeline and connect it */
gst_bin_add (GST_BIN (pipeline), webrtc);
sinkpad = gst_element_get_static_pad (q, "sink");
g_assert_nonnull (sinkpad);
ret = gst_pad_link (srcpad, sinkpad);
g_assert_cmpint (ret, ==, GST_PAD_LINK_OK);
gst_object_unref (srcpad);
@ -310,8 +357,8 @@ add_webrtcbin_to_pipeline (const gchar * peer_id, gboolean offer)
/* This is the gstwebrtc entry point where we create the offer and so on. It
* will be called when the pipeline goes to PLAYING.
* XXX: We must connect this after webrtcbin has been linked to a source via
* get_request_pad() otherwise webrtcbin will create an SDP offer with no
* media lines in it. */
* get_request_pad() and before we go from NULL->READY otherwise webrtcbin
* will create an SDP offer with no media lines in it. */
if (offer)
g_signal_connect (webrtc, "on-negotiation-needed",
G_CALLBACK (on_negotiation_needed), (gpointer) peer_id);
@ -325,22 +372,23 @@ add_webrtcbin_to_pipeline (const gchar * peer_id, gboolean offer)
g_signal_connect (webrtc, "pad-added", G_CALLBACK (on_incoming_stream),
pipeline);
/* Set to bin to PLAYING */
/* Set to pipeline branch to PLAYING */
ret = gst_element_sync_state_with_parent (q);
g_assert_true (ret);
ret = gst_element_sync_state_with_parent (webrtc);
gst_object_unref (webrtc);
g_assert_true (ret);
}
static void
call_peer (const gchar * peer_id)
{
add_webrtcbin_to_pipeline (peer_id, TRUE);
add_peer_to_pipeline (peer_id, TRUE);
}
static void
incoming_call_from_peer (const gchar * peer_id)
{
add_webrtcbin_to_pipeline (peer_id, FALSE);
add_peer_to_pipeline (peer_id, FALSE);
}
#define STR(x) #x
@ -755,6 +803,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
g_assert_nonnull (peer_id);
peers = g_list_remove (peers, peer_id);
g_print ("Peer %s has left the room\n", peer_id);
remove_peer_from_pipeline (peer_id);
g_free ((gchar*) peer_id);
/* TODO: cleanup pipeline */
} else {