}
}
-#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;
}
+
+#define STUN_SERVER " 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_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS"
+#define RTP_OPUS_DEFAULT_PT 97
+#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8"
+#define RTP_VP8_DEFAULT_PT 96
static gboolean
-start_pipeline (gboolean create_offer)
+start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
{
+ char *pipeline;
GstStateChangeReturn ret;
GError *error = NULL;
- pipe1 =
- gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv "
+ pipeline =
+ g_strdup_printf ("webrtcbin bundle-policy=max-bundle name=sendrecv "
STUN_SERVER
"videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! "
/* increase the default keyframe distance, browsers have really long
/* 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 ! "
- "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
+ "queue ! %s,payload=%u ! sendrecv. "
"audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! "
- "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);
+ "queue ! %s,payload=%u ! sendrecv. ", RTP_CAPS_VP8, vp8_pt,
+ RTP_CAPS_OPUS, opus_pt);
+ pipe1 = gst_parse_launch (pipeline, &error);
+ g_free (pipeline);
if (error) {
gst_printerr ("Failed to parse launch: %s\n", error->message);
g_error_free (error);
webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
g_assert_nonnull (webrtc1);
- if (remote_is_offerer) {
+ if (!create_offer) {
/* XXX: this will fail when the remote offers twcc as the extension id
* cannot currently be negotiated when receiving an offer.
*/
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");
webrtcbin name=sendrecv bundle-policy=max-bundle stun-server=stun://stun.l.google.com:19302
videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay picture-id-mode=15-bit !
- queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
+ queue ! application/x-rtp,media=video,encoding-name=VP8,payload={vp8_pt} ! sendrecv.
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
- queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
+ queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={opus_pt} ! sendrecv.
'''
from websockets.version import version as wsv
print(f'!!! {msg}', file=sys.stderr)
+def get_payload_types(sdpmsg, video_encoding, audio_encoding):
+ '''
+ Find the payload types for the specified video and audio encoding.
+
+ Very simplistically finds the first payload type matching the encoding
+ name. More complex applications will want to match caps on
+ profile-level-id, packetization-mode, etc.
+ '''
+ video_pt = None
+ audio_pt = None
+ for i in range(0, sdpmsg.medias_len()):
+ media = sdpmsg.get_media(i)
+ for j in range(0, media.formats_len()):
+ fmt = media.get_format(j)
+ if fmt == 'webrtc-datachannel':
+ continue
+ pt = int(fmt)
+ caps = media.get_caps_from_media(pt)
+ s = caps.get_structure(0)
+ encoding_name = s['encoding-name']
+ if video_pt is None and encoding_name == video_encoding:
+ video_pt = pt
+ elif audio_pt is None and encoding_name == audio_encoding:
+ audio_pt = pt
+ return {video_encoding: video_pt, audio_encoding: audio_pt}
+
+
class WebRTCClient:
def __init__(self, loop, our_id, peer_id, server, remote_is_offerer):
self.conn = None
print_status('Call was connected: creating offer')
promise = Gst.Promise.new_with_change_func(self.on_offer_created, None, None)
self.webrtc.emit('create-offer', None, promise)
- elif self.remote_is_offerer:
- # We are initiating the call, but we want the remote peer to create the offer
- print_status('Call was connected: requesting remote peer for offer')
- self.send_soon('OFFER_REQUEST')
def send_ice_candidate_message(self, _, mlineindex, candidate):
icemsg = json.dumps({'ice': {'candidate': candidate, 'sdpMLineIndex': mlineindex}})
decodebin.sync_state_with_parent()
self.webrtc.link(decodebin)
- def start_pipeline(self, create_offer=True):
+ def start_pipeline(self, create_offer=True, opus_pt=96, vp8_pt=97):
print_status(f'Creating pipeline, create_offer: {create_offer}')
- self.pipe = Gst.parse_launch(PIPELINE_DESC)
+ self.pipe = Gst.parse_launch(PIPELINE_DESC.format(vp8_pt=vp8_pt, opus_pt=opus_pt))
self.webrtc = self.pipe.get_by_name('sendrecv')
self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed, create_offer)
self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
self.webrtc.emit('create-answer', None, promise)
def handle_json(self, message):
- assert (self.webrtc)
try:
msg = json.loads(message)
except json.decoder.JSONDecoderError:
print_status('Received offer:\n%s' % sdp)
res, sdpmsg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(sdp.encode()), sdpmsg)
+
+ if not self.webrtc:
+ print_status('Incoming call: received an offer, creating pipeline')
+ pts = get_payload_types(sdpmsg, video_encoding='VP8', audio_encoding='OPUS')
+ assert('VP8' in pts)
+ assert('OPUS' in pts)
+ self.start_pipeline(create_offer=False, vp8_pt=pts['VP8'], opus_pt=pts['OPUS'])
+
+ assert(self.webrtc)
+
offer = GstWebRTC.WebRTCSessionDescription.new(GstWebRTC.WebRTCSDPType.OFFER, sdpmsg)
promise = Gst.Promise.new_with_change_func(self.on_offer_set, None, None)
self.webrtc.emit('set-remote-description', offer, promise)
elif 'ice' in msg:
+ assert(self.webrtc)
ice = msg['ice']
candidate = ice['candidate']
sdpmlineindex = ice['sdpMLineIndex']
self.pipe = None
self.webrtc = None
- def is_incoming_offer(self, msg):
- if self.webrtc:
- return False
- if self.remote_is_offerer:
- return True
- return True
-
async def loop(self):
assert self.conn
async for message in self.conn:
await self.setup_call()
elif message == 'SESSION_OK':
if self.remote_is_offerer:
- self.start_pipeline(create_offer=False)
+ # We are initiating the call, but we want the remote peer to create the offer
+ print_status('Call was connected: requesting remote peer for offer')
+ await self.send('OFFER_REQUEST')
else:
self.start_pipeline()
elif message == 'OFFER_REQUEST':
self.close_pipeline()
return 1
else:
- if self.is_incoming_offer(message):
- print_status('Incoming call: received an offer, creating pipeline')
- self.start_pipeline(create_offer=False)
self.handle_json(message)
self.close_pipeline()
return 0
projects / platform / upstream / gstreamer.git / commitdiff