webrtc_sendrecv.py: Add AV1 support when creating the offer

Requires svtav1enc at present for simplicity.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4644>

1 files changed, 26 insertions, 9 deletions

diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
index 5c794bfdcc..e3c12241f1 100755
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py
@@ -34,7 +34,7 @@ except ImportError:
 # These properties all mirror the ones in webrtc-sendrecv.c, see there for explanations
 PIPELINE_DESC_VP8 = '''
 webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
- videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
+ 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={video_pt} ! sendrecv.
  audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
@@ -42,14 +42,25 @@ webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
 '''
 PIPELINE_DESC_H264 = '''
 webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
- videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! \
-  x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true ! rtph264pay aggregate-mode=zero-latency config-interval=-1 !
+ videotestsrc is-live=true pattern=ball ! videoconvert ! queue !
+  x264enc tune=zerolatency speed-preset=ultrafast key-int-max=30 intra-refresh=true !
+  rtph264pay aggregate-mode=zero-latency config-interval=-1 !
   queue ! application/x-rtp,media=video,encoding-name=H264,payload={video_pt} ! sendrecv.
  audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
   queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
 '''
+# Force I420 because dav1d bundled with Chrome doesn't support 10-bit choma/luma (I420_10LE)
+PIPELINE_DESC_AV1 = '''
+webrtcbin name=sendrecv latency=0 stun-server=stun://stun.l.google.com:19302
+ videotestsrc is-live=true pattern=ball ! videoconvert ! queue !
+  video/x-raw,format=I420 ! svtav1enc preset=13 ! av1parse ! rtpav1pay !
+  queue ! application/x-rtp,media=video,encoding-name=AV1,payload={video_pt} ! sendrecv.
+ audiotestsrc is-live=true ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+  queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload={audio_pt} ! sendrecv.
+'''
 
 PIPELINE_DESC = {
+    'AV1': PIPELINE_DESC_AV1,
     'H264': PIPELINE_DESC_H264,
     'VP8': PIPELINE_DESC_VP8,
 }
@@ -334,22 +345,26 @@ class WebRTCClient:
         self.conn = None
 
 
-def check_plugins():
-    needed = ["opus", "vpx", "nice", "webrtc", "dtls", "srtp", "rtp",
+def check_plugins(video_encoding):
+    needed = ["opus", "nice", "webrtc", "dtls", "srtp", "rtp",
               "rtpmanager", "videotestsrc", "audiotestsrc"]
+    if video_encoding == 'vp8':
+        needed.append('vpx')
+    elif video_encoding == 'h264':
+        needed += ['x264', 'videoparsersbad']
+    elif video_encoding == 'av1':
+        needed += ['svtav1', 'videoparsersbad']
     missing = list(filter(lambda p: Gst.Registry.get().find_plugin(p) is None, needed))
     if len(missing):
-        print_error('Missing gstreamer plugins:', missing)
+        print_error(f'Missing gstreamer plugins: {missing}')
         return False
     return True
 
 
 if __name__ == '__main__':
     Gst.init(None)
-    if not check_plugins():
-        sys.exit(1)
     parser = argparse.ArgumentParser()
-    parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264'],
+    parser.add_argument('--video-encoding', default='vp8', nargs='?', choices=['vp8', 'h264', 'av1'],
                         help='Video encoding to negotiate')
     parser.add_argument('--peer-id', help='String ID of the peer to connect to')
     parser.add_argument('--our-id', help='String ID that the peer can use to connect to us')
@@ -359,6 +374,8 @@ if __name__ == '__main__':
                         dest='remote_is_offerer',
                         help='Request that the peer generate the offer and we\'ll answer')
     args = parser.parse_args()
+    if not check_plugins(args.video_encoding):
+        sys.exit(1)
     if not args.peer_id and not args.our_id:
         print('You must pass either --peer-id or --our-id')
         sys.exit(1)