#include <gst/sdp/sdp.h>
#include <gst/rtp/rtp.h>
-#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
+#include <gst/webrtc/nice/nice.h>
+
+#include "custom_agent.h"
/* For signalling */
#include <libsoup/soup.h>
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GMainLoop *loop;
-static GstElement *pipe1, *webrtc1 = NULL;
+static GstElement *pipe1, *webrtc1, *audio_bin, *video_bin = NULL;
static GObject *send_channel, *receive_channel;
static SoupWebsocketConnection *ws_conn = NULL;
static enum AppState app_state = 0;
static gchar *peer_id = NULL;
static gchar *our_id = NULL;
-static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
+static const gchar *server_url = "wss://webrtc.gstreamer.net:8443";
static gboolean disable_ssl = FALSE;
static gboolean remote_is_offerer = FALSE;
+static gboolean custom_ice = FALSE;
static GOptionEntry entries[] = {
{"peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id,
{"disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL},
{"remote-offerer", 0, 0, G_OPTION_ARG_NONE, &remote_is_offerer,
"Request that the peer generate the offer and we'll answer", NULL},
+ {"custom-ice", 0, 0, G_OPTION_ARG_NONE, &custom_ice,
+ "Use a custom ice agent", NULL},
{NULL},
};
}
}
-#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "
-#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
-#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="
-
static void
data_channel_on_error (GObject * dc, gpointer user_data)
{
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"
+#define RTP_OPUS_DEFAULT_PT 97
+#define RTP_VP8_DEFAULT_PT 96
static gboolean
-start_pipeline (gboolean create_offer)
+start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
{
+ GstBus *bus;
+ char *audio_desc, *video_desc;
GstStateChangeReturn ret;
- GError *error = NULL;
+ GstWebRTCICE *custom_agent;
+ GError *audio_error = NULL;
+ GError *video_error = NULL;
+
+ pipe1 = gst_pipeline_new ("webrtc-pipeline");
+
+ audio_desc =
+ g_strdup_printf
+ ("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
+ "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u "
+ "! application/x-rtp, encoding-name=OPUS ! queue", opus_pt);
+ audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
+ g_free (audio_desc);
+ if (audio_error) {
+ gst_printerr ("Failed to parse audio_bin: %s\n", audio_error->message);
+ g_error_free (audio_error);
+ goto err;
+ }
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
- STUN_SERVER
- "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
+ video_desc =
+ g_strdup_printf
+ ("videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
/* increase the default keyframe distance, browsers have really long
* periods between keyframes and rely on PLI events on packet loss to
* fix corrupted video.
*/
"vp8enc deadline=1 keyframe-max-dist=2000 ! "
- /* picture-id-mode=15-bit seems to make TWCC stats behave better */
- "rtpvp8pay name=videopay picture-id-mode=15-bit ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
- "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);
-
- if (error) {
- gst_printerr ("Failed to parse launch: %s\n", error->message);
- g_error_free (error);
+ /* picture-id-mode=15-bit seems to make TWCC stats behave better, and
+ * fixes stuttery video playback in Chrome */
+ "rtpvp8pay name=videopay picture-id-mode=15-bit pt=%u ! queue", vp8_pt);
+ video_bin = gst_parse_bin_from_description (video_desc, TRUE, &video_error);
+ g_free (video_desc);
+ if (video_error) {
+ gst_printerr ("Failed to parse video_bin: %s\n", video_error->message);
+ g_error_free (video_error);
goto err;
}
- webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
+ if (custom_ice) {
+ custom_agent = GST_WEBRTC_ICE (customice_agent_new ("custom"));
+ webrtc1 = gst_element_factory_make_full ("webrtcbin", "name", "sendrecv",
+ "stun-server", STUN_SERVER, "ice-agent", custom_agent, NULL);
+ } else {
+ webrtc1 = gst_element_factory_make_full ("webrtcbin", "name", "sendrecv",
+ "stun-server", STUN_SERVER, NULL);
+ }
g_assert_nonnull (webrtc1);
+ gst_util_set_object_arg (G_OBJECT (webrtc1), "bundle-policy", "max-bundle");
- if (remote_is_offerer) {
+ /* Takes ownership of each: */
+ gst_bin_add_many (GST_BIN (pipe1), audio_bin, video_bin, webrtc1, NULL);
+
+ if (!gst_element_link (audio_bin, webrtc1)) {
+ gst_printerr ("Failed to link audio_bin \n");
+ }
+ if (!gst_element_link (video_bin, webrtc1)) {
+ gst_printerr ("Failed to link video_bin \n");
+ }
+
+ if (!create_offer) {
/* XXX: this will fail when the remote offers twcc as the extension id
* cannot currently be negotiated when receiving an offer.
*/
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,
/* Incoming streams will be exposed via this signal */
g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
pipe1);
- /* Lifetime is the same as the pipeline itself */
- gst_object_unref (webrtc1);
g_timeout_add (100, (GSourceFunc) webrtcbin_get_stats, webrtc1);
GstWebRTCSessionDescription *offer = NULL;
GstPromise *promise;
+ /* If we got an offer and we have no webrtcbin, we need to parse the SDP,
+ * get the payload types, then start the pipeline */
+ if (!webrtc1 && our_id) {
+ guint medias_len, formats_len;
+ guint opus_pt = 0, vp8_pt = 0;
+
+ gst_println ("Parsing offer to find payload types");
+
+ medias_len = gst_sdp_message_medias_len (sdp);
+ for (int i = 0; i < medias_len; i++) {
+ const GstSDPMedia *media = gst_sdp_message_get_media (sdp, i);
+ formats_len = gst_sdp_media_formats_len (media);
+ for (int j = 0; j < formats_len; j++) {
+ guint pt;
+ GstCaps *caps;
+ GstStructure *s;
+ const char *fmt, *encoding_name;
+
+ fmt = gst_sdp_media_get_format (media, j);
+ if (g_strcmp0 (fmt, "webrtc-datachannel") == 0)
+ continue;
+ pt = atoi (fmt);
+ caps = gst_sdp_media_get_caps_from_media (media, pt);
+ s = gst_caps_get_structure (caps, 0);
+ encoding_name = gst_structure_get_string (s, "encoding-name");
+ if (vp8_pt == 0 && g_strcmp0 (encoding_name, "VP8") == 0)
+ vp8_pt = pt;
+ if (opus_pt == 0 && g_strcmp0 (encoding_name, "OPUS") == 0)
+ opus_pt = pt;
+ }
+ }
+
+ g_assert_cmpint (opus_pt, !=, 0);
+ g_assert_cmpint (vp8_pt, !=, 0);
+
+ gst_println ("Starting pipeline with opus pt: %u vp8 pt: %u", opus_pt,
+ vp8_pt);
+
+ if (!start_pipeline (FALSE, opus_pt, vp8_pt)) {
+ cleanup_and_quit_loop ("ERROR: failed to start pipeline",
+ PEER_CALL_ERROR);
+ }
+ }
+
offer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, sdp);
g_assert_nonnull (offer);
app_state = PEER_CONNECTED;
/* Start negotiation (exchange SDP and ICE candidates) */
- if (!start_pipeline (TRUE))
+ if (!start_pipeline (TRUE, RTP_OPUS_DEFAULT_PT, RTP_VP8_DEFAULT_PT))
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
} else if (g_strcmp0 (text, "OFFER_REQUEST") == 0) {
}
gst_print ("Received OFFER_REQUEST, sending offer\n");
/* Peer wants us to start negotiation (exchange SDP and ICE candidates) */
- if (!start_pipeline (TRUE))
+ if (!start_pipeline (TRUE, RTP_OPUS_DEFAULT_PT, RTP_VP8_DEFAULT_PT))
cleanup_and_quit_loop ("ERROR: failed to start pipeline",
PEER_CALL_ERROR);
} else if (g_str_has_prefix (text, "ERROR")) {
goto out;
}
- /* If peer connection wasn't made yet and we are expecting peer will
- * connect to us, launch pipeline at this moment */
- if (!webrtc1 && our_id) {
- if (!start_pipeline (FALSE)) {
- cleanup_and_quit_loop ("ERROR: failed to start pipeline",
- PEER_CALL_ERROR);
- }
-
- app_state = PEER_CALL_NEGOTIATING;
- }
-
object = json_node_get_object (root);
/* Check type of JSON message */
if (json_object_has_member (object, "sdp")) {
const gchar *text, *sdptype;
GstWebRTCSessionDescription *answer;
- g_assert_cmphex (app_state, ==, PEER_CALL_NEGOTIATING);
+ app_state = PEER_CALL_NEGOTIATING;
child = json_object_get_object_member (object, "sdp");
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);
}
projects / platform / upstream / gstreamer.git / blobdiff