Mercurial > libervia-backend

diff libervia/frontends/tools/webrtc_models.py @ 4240:79c8a70e1813

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backend, frontend: prepare remote control: This is a series of changes necessary to prepare the implementation of remote control feature: - XEP-0166: add a `priority` attribute to `ApplicationData`: this is needed when several applications are working in a same session, to know which one must be handled first. Will be used to make Remote Control have precedence over Call content. - XEP-0166: `_call_plugins` is now async and is not used with `DeferredList` anymore: the benefit to have methods called in parallels is very low, and it cause a lot of trouble as we can't predict order. Methods are now called sequentially so workflow can be predicted. - XEP-0167: fix `senders` XMPP attribute <=> SDP mapping - XEP-0234: preflight acceptance key is now `pre-accepted` instead of `file-accepted`, so the same key can be used with other jingle applications. - XEP-0167, XEP-0343: move some method to XEP-0167 - XEP-0353: use new `priority` feature to call preflight methods of applications according to it. - frontend (webrtc): refactor the sources/sink handling with a more flexible mechanism based on Pydantic models. It is now possible to have has many Data Channel as necessary, to have them in addition to A/V streams, to specify manually GStreamer sources and sinks, etc. - frontend (webrtc): rework of the pipeline to reduce latency. - frontend: new `portal_desktop` method. Screenshare portal handling has been moved there, and RemoteDesktop portal has been added. - frontend (webrtc): fix `extract_ufrag_pwd` method. rel 436
author Goffi <goffi@goffi.org>
date Sat, 11 May 2024 13:52:41 +0200
parents d01b8d002619
children 0d7bb4df2343
line wrap: on
line diff
--- a/libervia/frontends/tools/webrtc_models.py	Sat May 11 13:25:45 2024 +0200
+++ b/libervia/frontends/tools/webrtc_models.py	Sat May 11 13:52:41 2024 +0200
@@ -16,8 +16,15 @@
 # You should have received a copy of the GNU Affero General Public License
 # along with this program.  If not, see <http://www.gnu.org/licenses/>.
 
+from collections.abc import Awaitable
 from dataclasses import dataclass, field
 from typing import Any, Callable
+import uuid
+import gi
+
+gi.require_versions({"Gst": "1.0", "GstWebRTC": "1.0"})
+from gi.repository import GstWebRTC
+from pydantic import BaseModel, Field
 
 from libervia.frontends.tools import jid
 
@@ -30,7 +37,80 @@
     kwargs: dict[str, Any] = field(default_factory=dict)
 
 
-@dataclass
-class AppSinkData:
+class SourcesData(BaseModel):
+    """Data for Sources"""
+
+
+class SourcesNone(SourcesData):
+    """No source is used.
+
+    This is used when the WebRTC connection will be used for data channels only."""
+
+
+class SourcesAuto(SourcesData):
+    """Automatic Sources (webcam/microphone)"""
+
+
+class SourcesTest(SourcesData):
+    """Test Sources (pattern)"""
+
+
+class SourcesDataChannel(SourcesData):
+    """Sources for transmitting data over Data Channel
+
+
+    @param dc_open_cb: Called when Data Channel is open.
+        This callback will be run in a GStreamer thread.
+    """
+    name: str = Field(default_factory=lambda: str(uuid.uuid4()))
+    dc_open_cb: Callable[[GstWebRTC.WebRTCDataChannel], None]
+
+
+class SourcesPipeline(SourcesData):
+    """Use custom pipeline description as a source.
+
+    @param video_pipeline: pipeline description of video source.
+        None to use automatic video source (same as SourcesAuto).
+        Empty string to disable video.
+    @param audio_pipeline: pipeline description of audio source.
+        None to use automatic audio source (same as SourcesAuto).
+        Empty string to disable audio.
+    @param video_properties: Elements properties to set.
+    @param audio_properties: Elements properties to set.
+
+    """
+    video_pipeline: str|None = None
+    audio_pipeline: str|None = None
+    video_properties: dict = Field(default_factory=lambda: {})
+    audio_properties: dict = Field(default_factory=lambda: {})
+
+
+class SinksData(BaseModel):
+    """Data for Sinks"""
+
+
+class SinksNone(SinksData):
+    """No sink is used.
+
+    This is used when the WebRTC connection will be used for data channels only."""
+
+
+class SinksAuto(SinksData):
+    """Automatic Sinks (create windows/default audio)"""
+
+
+class SinksApp(SinksData):
     local_video_cb: Callable
-    remote_video_cb: Callable|None
+    remote_video_cb: Callable | None
+
+
+class SinksDataChannel(SinksData):
+    """Sinks for transmitting data over Data Channel
+
+    @param dc_on_data_channel: Called when Data Channel is created.
+        This callback will be run in a GStreamer thread.
+    """
+
+    dc_on_data_channel: (
+        Callable[[GstWebRTC.WebRTCDataChannel], Awaitable[None]] | None
+    ) = None