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examples: webrtc: Add bus handling to the Android and C sendrecv examples

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Without a bus, messages will just pile up and errors are not handled at
all. Also without handling the LATENCY messages the latency configured
on the pipeline will be wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3609>
This commit is contained in:
Sebastian Dröge 2022-12-20 13:02:08 +02:00
parent 9914ff9b4c
commit d10981f7b9
2 changed files with 100 additions and 0 deletions
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50
subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c
View file

@ -107,7 +107,14 @@ cleanup_and_quit_loop (WebRTC * webrtc, const gchar * msg, enum AppState state)
}
if (webrtc->pipe) {
GstBus *bus;
gst_element_set_state (webrtc->pipe, GST_STATE_NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (webrtc->pipe));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
gst_object_unref (webrtc->pipe);
webrtc->pipe = NULL;
}
@ -328,6 +335,45 @@ add_fec_to_offer (GstElement * webrtc)
"fec-percentage", 25, "do-nack", FALSE, NULL);
}
static gboolean
bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
WebRTC *webrtc = user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_error (message, &error, &debug);
cleanup_and_quit_loop (webrtc, "ERROR: error on bus", APP_STATE_ERROR);
g_warning ("Error on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_WARNING:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_warning (message, &error, &debug);
g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_LATENCY:
gst_bin_recalculate_latency (GST_BIN (webrtc->pipe));
break;
default:
break;
}
return G_SOURCE_CONTINUE;
}
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload=100"
#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload=101"
@ -352,6 +398,10 @@ start_pipeline (WebRTC * webrtc)
goto err;
}
bus = gst_pipeline_get_bus (GST_PIPELINE (webrtc->pipe));
gst_bus_add_watch (bus, bus_watch_cb, webrtc);
gst_object_unref (bus);
webrtc->webrtcbin = gst_bin_get_by_name (GST_BIN (webrtc->pipe), "sendrecv");
g_assert (webrtc->webrtcbin != NULL);
add_fec_to_offer (webrtc->webrtcbin);

50
subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
View file

@ -422,6 +422,44 @@ webrtcbin_get_stats (GstElement * webrtcbin)
return G_SOURCE_REMOVE;
}
static gboolean
bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
GstPipeline *pipeline = user_data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_error (message, &error, &debug);
cleanup_and_quit_loop ("ERROR: Error on bus", APP_STATE_ERROR);
g_warning ("Error on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_WARNING:
{
GError *error = NULL;
gchar *debug = NULL;
gst_message_parse_warning (message, &error, &debug);
g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
g_error_free (error);
g_free (debug);
break;
}
case GST_MESSAGE_LATENCY:
gst_bin_recalculate_latency (GST_BIN (pipeline));
break;
default:
break;
}
return G_SOURCE_CONTINUE;
}
#define STUN_SERVER "stun://stun.l.google.com:19302"
#define RTP_TWCC_URI "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
@ -431,6 +469,7 @@ webrtcbin_get_stats (GstElement * webrtcbin)
static gboolean
start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
{
GstBus *bus;
char *audio_desc, *video_desc;
GstStateChangeReturn ret;
GstWebRTCICE *custom_agent;
@ -532,6 +571,10 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
g_signal_connect (webrtc1, "notify::ice-gathering-state",
G_CALLBACK (on_ice_gathering_state_notify), NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
gst_bus_add_watch (bus, bus_watch_cb, pipe1);
gst_object_unref (bus);
gst_element_set_state (pipe1, GST_STATE_READY);
g_signal_emit_by_name (webrtc1, "create-data-channel", "channel", NULL,
@ -1029,8 +1072,15 @@ main (int argc, char *argv[])
g_main_loop_unref (loop);
if (pipe1) {
GstBus *bus;
gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
gst_print ("Pipeline stopped\n");
bus = gst_pipeline_get_bus (GST_PIPELINE (pipe1));
gst_bus_remove_watch (bus);
gst_object_unref (bus);
gst_object_unref (pipe1);
}