streamlit-webrtc 0.53.10__tar.gz → 0.54.0__tar.gz

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/PKG-INFO

@@ -1,13 +1,13 @@
 Metadata-Version: 2.4
 Name: streamlit-webrtc
-Version: 0.53.10
+Version: 0.54.0
 Summary: Real-time video and audio processing on Streamlit
 Project-URL: Repository, https://github.com/whitphx/streamlit-webrtc
 Author-email: "Yuichiro Tachibana (Tsuchiya)" <t.yic.yt@gmail.com>
 License-Expression: MIT
 License-File: LICENSE
 Requires-Python: !=3.9.7,>=3.9
-Requires-Dist: aiortc>=1.9.0
+Requires-Dist: aiortc>=1.11.0
 Requires-Dist: packaging>=20.0
 Requires-Dist: streamlit>=0.84.1
 Description-Content-Type: text/markdown

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/pyproject.toml

@@ -10,7 +10,7 @@ dependencies = [
     # For allow-same-origin, >=0.73.0 is required. See https://blog.streamlit.io/streamlit-components-security-and-a-five-month-quest-to-ship-a-single-line-of-code/
     # 0.84.0 has an error at marshalling component values.
     "streamlit>=0.84.1",
-    "aiortc>=1.9.0",  # aiortc<1.4.0 causes an error with cryptography>=39.0.0. See https://github.com/whitphx/streamlit-webrtc/issues/1164. The fix was introduced into aiortc in https://github.com/aiortc/aiortc/commit/08b0a7e9f5030a9f7e5617382e92560d4ae763a2 that 1.4.0 included.
+    "aiortc>=1.11.0",  # aiortc<1.4.0 causes an error with cryptography>=39.0.0. See https://github.com/whitphx/streamlit-webrtc/issues/1164. The fix was introduced into aiortc in https://github.com/aiortc/aiortc/commit/08b0a7e9f5030a9f7e5617382e92560d4ae763a2 that 1.4.0 included.
     "packaging>=20.0",
 ]
 

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/streamlit_webrtc/_compat.py

@@ -56,8 +56,10 @@ except ModuleNotFoundError:
         from streamlit.app_session import AppSession, AppSessionState  # type: ignore
     except ModuleNotFoundError:
         # streamlit < 1.4
-        from streamlit.report_session import (  # type: ignore # isort:skip
+        from streamlit.report_session import (  # type: ignore
             ReportSession as AppSession,
+        )
+        from streamlit.report_session import (  # type: ignore
             ReportSessionState as AppSessionState,
         )
 

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/streamlit_webrtc/session_info.py

@@ -34,11 +34,11 @@ def get_this_session_info() -> Optional[SessionInfo]:
     if VER_GTE_1_18_0:
         from streamlit.runtime.runtime import Runtime
 
-        return Runtime.instance()._session_mgr.get_session_info(session_id)  # type: ignore  # noqa: E501
+        return Runtime.instance()._session_mgr.get_session_info(session_id)  # type: ignore
     elif VER_GTE_1_14_0:
         from streamlit.runtime.runtime import Runtime
 
-        return Runtime.instance()._get_session_info(session_id)  # type: ignore  # noqa: E501
+        return Runtime.instance()._get_session_info(session_id)  # type: ignore
 
     current_server = get_current_server()
 

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/streamlit_webrtc/shutdown.py

@@ -40,7 +40,7 @@ class SessionShutdownObserver:
     ):
         # Use polling because event-based methods are not available
         # to observe the session lifecycle.
-        while not self._polling_thread_stop_event.wait(1.0):
+        while True:
             app_session = app_session_ref()
             if not app_session:
                 logger.debug("AppSession has removed.")
@@ -51,6 +51,11 @@ class SessionShutdownObserver:
                     app_session.id,
                 )
                 break
+            if self._polling_thread_stop_event.wait(1.0):
+                logger.debug(
+                    "The polling thread should be stopped. Exit the polling loop and return."
+                )
+                return
 
         # Ensure the flag is set
         self._polling_thread_stop_event.set()

{streamlit_webrtc-0.53.10 → streamlit_webrtc-0.54.0}/streamlit_webrtc/webrtc.py

@@ -4,7 +4,7 @@ import itertools
 import logging
 import queue
 import threading
-from typing import Callable, Generic, Literal, Optional, Union, cast
+from typing import TYPE_CHECKING, Callable, Generic, Literal, Optional, Union, cast
 
 from aiortc import RTCConfiguration, RTCPeerConnection, RTCSessionDescription
 from aiortc.contrib.media import MediaPlayer, MediaRecorder, MediaRelay
@@ -39,6 +39,9 @@ from .process import (
 from .receive import AudioReceiver, VideoReceiver
 from .relay import get_global_relay
 
+if TYPE_CHECKING:
+    import concurrent.futures
+
 __all__ = [
     "AudioProcessorBase",
     "AudioProcessorFactory",
@@ -48,7 +51,7 @@ __all__ = [
     "MediaRecorderFactory",
     "VideoProcessorFactory",
     "WebRtcMode",
-    "TimeoutError",
+    "SignallingTimeoutError",
     "WebRtcWorker",
 ]
 
@@ -63,7 +66,7 @@ class WebRtcMode(enum.Enum):
     SENDRECV = enum.auto()
 
 
-class TimeoutError(Exception):
+class SignallingTimeoutError(Exception):
     pass
 
 
@@ -522,7 +525,7 @@ class WebRtcWorker(Generic[VideoProcessorT, AudioProcessorT]):
             if self.pc.iceConnectionState == "failed":
                 await self.pc.close()
 
-        process_offer_task = loop.create_task(
+        process_offer_task = asyncio.run_coroutine_threadsafe(
             _process_offer_coro(
                 self.mode,
                 self.pc,
@@ -540,10 +543,11 @@ class WebRtcWorker(Generic[VideoProcessorT, AudioProcessorT]):
                 sendback_video=self.sendback_video,
                 sendback_audio=self.sendback_audio,
                 on_track_created=on_track_created,
-            )
+            ),
+            loop=loop,
         )
 
-        def callback(done_task: asyncio.Task):
+        def callback(done_task: "concurrent.futures.Future"):
             e = done_task.exception()
             if e:
                 logger.debug("Error occurred in process_offer")
@@ -574,7 +578,7 @@ class WebRtcWorker(Generic[VideoProcessorT, AudioProcessorT]):
             result = self._answer_queue.get(block=True, timeout=timeout)
         except queue.Empty:
             self.stop(timeout=1)
-            raise TimeoutError(
+            raise SignallingTimeoutError(
                 "Processing offer and initializing the worker "
                 f"has not finished in {timeout} seconds"
             )
@@ -672,6 +676,8 @@ class WebRtcWorker(Generic[VideoProcessorT, AudioProcessorT]):
         self._relayed_source_video_track = None
 
     def stop(self, timeout: Union[float, None] = 1.0):
+        logger.debug("Stopping the WebRTC worker")
+
         self._unset_processors()
         if self._process_offer_thread:
             self._process_offer_thread.join(timeout=timeout)
@@ -680,7 +686,7 @@ class WebRtcWorker(Generic[VideoProcessorT, AudioProcessorT]):
         if self.pc and self.pc.connectionState != "closed":
             loop = get_global_event_loop()
             if loop.is_running():
-                loop.create_task(self.pc.close())
+                asyncio.run_coroutine_threadsafe(self.pc.close(), loop=loop)
             else:
                 loop.run_until_complete(self.pc.close())