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webrtc sendonly: Add priority to example

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/18>
This commit is contained in:
Olivier Crête 2020-07-09 17:07:10 -04:00
parent 992cb3c5f4
commit 4e141f1076
1 changed files with 64 additions and 6 deletions
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70
webrtc/sendonly/webrtc-unidirectional-h264.c
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@ -15,17 +15,19 @@
#include <string.h>
#define RTP_PAYLOAD_TYPE "96"
#define RTP_AUDIO_PAYLOAD_TYPE "97"
#define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302"
gchar *video_priority = NULL;
gchar *audio_priority = NULL;
typedef struct _ReceiverEntry ReceiverEntry;
ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
void destroy_receiver_entry (gpointer receiver_entry_ptr);
GstPadProbeReturn payloader_caps_event_probe_cb (GstPad * pad,
GstPadProbeInfo * info, gpointer user_data);
void on_offer_created_cb (GstPromise * promise, gpointer user_data);
void on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data);
void on_ice_candidate_cb (GstElement * webrtcbin, guint mline_index,
@ -191,6 +193,24 @@ bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
return G_SOURCE_CONTINUE;
}
static GstWebRTCPriorityType
_priority_from_string (const gchar * s)
{
GEnumClass *klass =
(GEnumClass *) g_type_class_ref (GST_TYPE_WEBRTC_PRIORITY_TYPE);
GEnumValue *en;
g_return_val_if_fail (klass, 0);
if (!(en = g_enum_get_value_by_name (klass, s)))
en = g_enum_get_value_by_nick (klass, s);
g_type_class_unref (klass);
if (en)
return en->value;
return 0;
}
ReceiverEntry *
create_receiver_entry (SoupWebsocketConnection * connection)
{
@ -215,7 +235,9 @@ create_receiver_entry (SoupWebsocketConnection * connection)
"v4l2src ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
"rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
"application/x-rtp,media=video,encoding-name=H264,payload="
RTP_PAYLOAD_TYPE " ! webrtcbin. ", &error);
RTP_PAYLOAD_TYPE " ! webrtcbin. "
"autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! opusenc ! rtpopuspay pt="
RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
if (error != NULL) {
g_error ("Could not create WebRTC pipeline: %s\n", error->message);
g_error_free (error);
@ -228,9 +250,25 @@ create_receiver_entry (SoupWebsocketConnection * connection)
g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",
&transceivers);
g_assert (transceivers != NULL && transceivers->len > 0);
g_assert (transceivers != NULL && transceivers->len > 1);
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
if (video_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (video_priority);
if (priority)
gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
}
trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
if (audio_priority) {
GstWebRTCPriorityType priority;
priority = _priority_from_string (audio_priority);
if (priority)
gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
}
g_array_unref (transceivers);
g_signal_connect (receiver_entry->webrtcbin, "on-negotiation-needed",
@ -583,15 +621,35 @@ exit_sighandler (gpointer user_data)
}
#endif
static GOptionEntry entries[] = {
{"video-priority", 0, 0, G_OPTION_ARG_STRING, &video_priority,
"Priority of the video stream (very-low, low, medium or high)",
"PRIORITY"},
{"audio-priority", 0, 0, G_OPTION_ARG_STRING, &audio_priority,
"Priority of the audio stream (very-low, low, medium or high)",
"PRIORITY"},
{NULL},
};
int
main (int argc, char *argv[])
{
GMainLoop *mainloop;
SoupServer *soup_server;
GHashTable *receiver_entry_table;
GOptionContext *context;
GError *error = NULL;
setlocale (LC_ALL, "");
gst_init (&argc, &argv);
context = g_option_context_new ("- gstreamer webrtc sendonly demo");
g_option_context_add_main_entries (context, entries, NULL);
g_option_context_add_group (context, gst_init_get_option_group ());
if (!g_option_context_parse (context, &argc, &argv, &error)) {
g_printerr ("Error initializing: %s\n", error->message);
return -1;
}
receiver_entry_table =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,