code

.gitignore

@@ -11,4 +11,6 @@ __tmp/*
 node_modules
 backend/**/templates/
 demo/MobileNetSSD_deploy.caffemodel
-demo/MobileNetSSD_deploy.prototxt.txt
+demo/MobileNetSSD_deploy.prototxt.txt
+.DS_Store
+test/

README.md

@@ -1,14 +1,3 @@
----
-tags: [gradio-custom-component, Video, streaming, webrtc, realtime]
-title: gradio_webrtc
-short_description: Stream images in realtime with webrtc
-colorFrom: blue
-colorTo: yellow
-sdk: gradio
-pinned: false
-app_file: space.py
----
-
 <h1 style='text-align: center; margin-bottom: 1rem'> Gradio WebRTC ⚡️ </h1>
 
 <div style="display: flex; flex-direction: row; justify-content: center">
@@ -30,15 +19,15 @@ pip install gradio_webrtc
 1. [Object Detection from Webcam with YOLOv10](https://huggingface.co/spaces/freddyaboulton/webrtc-yolov10n) 📷
 2. [Streaming Object Detection from Video with RT-DETR](https://huggingface.co/spaces/freddyaboulton/rt-detr-object-detection-webrtc) 🎥
 3. [Text-to-Speech](https://huggingface.co/spaces/freddyaboulton/parler-tts-streaming-webrtc) 🗣️
+4. [Conversational AI](https://huggingface.co/spaces/freddyaboulton/omni-mini-webrtc) 🤖🗣️
 
 ## Usage
 
 The WebRTC component supports the following three use cases:
-1. Streaming video from the user webcam to the server and back
+1. [Streaming video from the user webcam to the server and back](#h-streaming-video-from-the-user-webcam-to-the-server-and-back)
-2. Streaming Video from the server to the client
+2. [Streaming Video from the server to the client](#h-streaming-video-from-the-server-to-the-client)
-3. Streaming Audio from the server to the client
+3. [Streaming Audio from the server to the client](#h-streaming-audio-from-the-server-to-the-client)
-
+4. [Streaming Audio from the client to the server and back (conversational AI)](#h-conversational-ai)
-Streaming Audio from client to the server and back (conversational AI) is not supported yet.
 
 
 ## Streaming Video from the User Webcam to the Server and Back
@@ -78,7 +67,7 @@ as a **numpy array** and returns the processed frame also as a **numpy array**.
 * The `inputs` parameter should be a list where the first element is the WebRTC component. The only output allowed is the WebRTC component.
 * The `time_limit` parameter is the maximum time in seconds the video stream will run. If the time limit is reached, the video stream will stop.
 
-## Streaming Video from the User Webcam to the Server and Back
+## Streaming Video from the server to the client
 
 ```python
 import gradio as gr
@@ -143,6 +132,52 @@ with gr.Blocks() as demo:
 * The numpy array should be of shape (1, num_samples).
 * The `outputs` parameter should be a list with the WebRTC component as the only element.
 
+## Conversational AI
+
+```python
+import gradio as gr
+import numpy as np
+from gradio_webrtc import WebRTC, StreamHandler
+from queue import Queue
+import time
+
+
+class EchoHandler(StreamHandler):
+    def __init__(self) -> None:
+        super().__init__()
+        self.queue = Queue()
+
+    def receive(self, frame: tuple[int, np.ndarray] | np.ndarray) -> None:
+        self.queue.put(frame)
+
+    def emit(self) -> None:
+        return self.queue.get()
+
+
+with gr.Blocks() as demo:
+    with gr.Column():
+        with gr.Group():
+            audio = WebRTC(
+                label="Stream",
+                rtc_configuration=None,
+                mode="send-receive",
+                modality="audio",
+            )
+
+        audio.stream(fn=EchoHandler(), inputs=[audio], outputs=[audio], time_limit=15)
+
+
+if __name__ == "__main__":
+    demo.launch()
+```
+
+* Instead of passing a function to the `stream` event's `fn` parameter, pass a `StreamHandler` implementation. The `StreamHandler` above simply echoes the audio back to the client.
+* The `StreamHandler` class has two methods: `receive` and `emit`. The `receive` method is called when a new frame is received from the client, and the `emit` method returns the next frame to send to the client.
+* An audio frame is represented as a tuple of (frame_rate, audio_samples) where `audio_samples` is a numpy array of shape (num_channels, num_samples).
+* You can also specify the audio layout ("mono" or "stereo") in the emit method by retuning it as the third element of the tuple. If not specified, the default is "mono".
+* The `time_limit` parameter is the maximum time in seconds the conversation will run. If the time limit is reached, the audio stream will stop.
+
+
 ## Deployment
 
 When deploying in a cloud environment (like Hugging Face Spaces, EC2, etc), you need to set up a TURN server to relay the WebRTC traffic.

backend/gradio_webrtc/utils.py

@@ -58,5 +58,5 @@ def player_worker_decode(
             frame.pts = audio_samples
             frame.time_base = audio_time_base
             audio_samples += frame.samples
-
             asyncio.run_coroutine_threadsafe(queue.put(frame), loop)
+            logger.debug("Queue size utils.py: %s", queue.qsize())

backend/gradio_webrtc/webrtc.py

@@ -99,9 +99,9 @@ class VideoCallback(VideoStreamTrack):
 
             return new_frame
         except Exception as e:
-            logger.debug(e)
+            logger.debug("exception %s", e)
             exec = traceback.format_exc()
-            logger.debug(exec)
+            logger.debug("traceback %s", exec)
 
 
 class StreamHandler(ABC):
@@ -161,20 +161,19 @@ class AudioCallback(AudioStreamTrack):
                 frame = cast(AudioFrame, await self.track.recv())
                 for frame in self.event_handler.resample(frame):
                     numpy_array = frame.to_ndarray()
-                    logger.debug("numpy array shape %s", numpy_array.shape)
                     await anyio.to_thread.run_sync(
                         self.event_handler.receive, (frame.sample_rate, numpy_array)
                     )
-            except MediaStreamError as e:
+            except MediaStreamError:
-                print("MediaStreamError", e)
+               logger.debug("MediaStreamError in process_input_frames")
-                break
+               break
 
     def start(self):
         if not self.has_started:
             asyncio.create_task(self.process_input_frames())
             self.__thread = threading.Thread(
-                name="audio-output-decoders",
                 target=player_worker_decode,
+                daemon=False,
                 args=(
                     asyncio.get_event_loop(),
                     self.event_handler.emit,
@@ -214,11 +213,12 @@ class AudioCallback(AudioStreamTrack):
             self.last_timestamp = time.time()
             return frame
         except Exception as e:
-            logger.debug(e)
+            logger.debug("exception %s", e)
             exec = traceback.format_exc()
-            logger.debug(exec)
+            logger.debug("traceback %s", exec)
 
     def stop(self):
+        logger.debug("audio callback stop")
         self.thread_quit.set()
         if self.__thread is not None:
             self.__thread.join()
@@ -266,9 +266,9 @@ class ServerToClientVideo(VideoStreamTrack):
             next_frame.time_base = time_base
             return next_frame
         except Exception as e:
-            logger.debug(e)
+            logger.debug("exception %s", e)
             exec = traceback.format_exc()
-            logger.debug(exec)
+            logger.debug("traceback %s ", exec)
 
 
 class ServerToClientAudio(AudioStreamTrack):
@@ -298,13 +298,14 @@ class ServerToClientAudio(AudioStreamTrack):
                 frame = next(self.generator)
                 return frame
             except StopIteration:
-                pass
+                self.thread_quit.set()
 
     def start(self):
         if self.__thread is None:
             self.__thread = threading.Thread(
                 name="generator-runner",
                 target=player_worker_decode,
+                daemon=True,
                 args=(
                     asyncio.get_event_loop(),
                     self.next,
@@ -338,9 +339,9 @@ class ServerToClientAudio(AudioStreamTrack):
 
             return data
         except Exception as e:
-            logger.debug(e)
+            logger.debug("exception %s", e)
             exec = traceback.format_exc()
-            logger.debug(exec)
+            logger.debug("traceback %s", exec)
 
     def stop(self):
         self.thread_quit.set()
@@ -606,9 +607,12 @@ class WebRTC(Component):
 
         @pc.on("connectionstatechange")
         async def on_connectionstatechange():
+            logger.debug("pc.connectionState %s", pc.connectionState)
             if pc.connectionState in ["failed", "closed"]:
                 await pc.close()
-                self.connections.pop(body["webrtc_id"], None)
+                connection = self.connections.pop(body["webrtc_id"], None)
+                if connection:
+                    connection.stop()
                 self.pcs.discard(pc)
             if pc.connectionState == "connected":
                 if self.time_limit is not None:

demo/app.py

@@ -57,7 +57,7 @@ pip install gradio_webrtc
 1. [Object Detection from Webcam with YOLOv10](https://huggingface.co/spaces/freddyaboulton/webrtc-yolov10n) 📷
 2. [Streaming Object Detection from Video with RT-DETR](https://huggingface.co/spaces/freddyaboulton/rt-detr-object-detection-webrtc) 🎥
 3. [Text-to-Speech](https://huggingface.co/spaces/freddyaboulton/parler-tts-streaming-webrtc) 🗣️
-4. [Conversational AI]()
+4. [Conversational AI](https://huggingface.co/spaces/freddyaboulton/omni-mini-webrtc) 🤖🗣️
 
 ## Usage
 

frontend/shared/InteractiveAudio.svelte

@@ -36,7 +36,7 @@
     let stream_state: "open" | "closed" | "waiting" = "closed";
     let audio_player: HTMLAudioElement;
     let pc: RTCPeerConnection;
-    let _webrtc_id = Math.random().toString(36).substring(2);
+    let _webrtc_id = null;
 
 
     const dispatch = createEventDispatcher<{
@@ -63,6 +63,7 @@
             _time_limit = null;
             return;
         }
+            _webrtc_id = Math.random().toString(36).substring(2);
             value = _webrtc_id;
             pc = new RTCPeerConnection(rtc_configuration);
             pc.addEventListener("connectionstatechange",

frontend/shared/StaticAudio.svelte

@@ -47,6 +47,7 @@
     async function start_stream(value: string): Promise<string> {
         if( value === "start_webrtc_stream") {
             stream_state = "waiting";
+            _webrtc_id = Math.random().toString(36).substring(2)
             value = _webrtc_id;
             console.log("set value to ", value);
             pc = new RTCPeerConnection(rtc_configuration);

frontend/shared/StaticVideo.svelte

@@ -40,6 +40,7 @@
 	)
 
 	$: if( value === "start_webrtc_stream") {
+		_webrtc_id = Math.random().toString(36).substring(2);
 		value = _webrtc_id;
 		pc = new RTCPeerConnection(rtc_configuration);
 		pc.addEventListener("connectionstatechange",

frontend/shared/Webcam.svelte

@@ -138,7 +138,7 @@
                 }
             )
             stream_state = "waiting"
-			webrtc_id = _webrtc_id;
+			webrtc_id = Math.random().toString(36).substring(2);
             start(stream, pc, video_source, server.offer, webrtc_id).then((connection) => {
 				pc = connection;
 			}).catch(() => {

frontend/shared/webrtc_utils.ts

@@ -134,6 +134,7 @@ export function stop(pc: RTCPeerConnection) {
 	// close local audio / video
 	if (pc.getSenders()) {
 		pc.getSenders().forEach((sender) => {
+			console.log("sender", sender);
 			if (sender.track && sender.track.stop) sender.track.stop();
 		});
 	}

pyproject.toml

@@ -8,7 +8,7 @@ build-backend = "hatchling.build"
 
 [project]
 name = "gradio_webrtc"
-version = "0.0.5"
+version = "0.0.6a2"
 description = "Stream images in realtime with webrtc"
 readme = "README.md"
 license = "apache-2.0"