Video Bugfix + generator (#96)

* Code

* Fix demo

* move to init

---------

Co-authored-by: Freddy Boulton <freddyboulton@hf-freddy.local>

backend/fastrtc/__init__.py

@@ -25,6 +25,7 @@ from .utils import (
     audio_to_bytes,
     audio_to_file,
     audio_to_float32,
+    audio_to_int16,
     wait_for_item,
 )
 from .webrtc import (
@@ -43,6 +44,7 @@ __all__ = [
     "audio_to_bytes",
     "audio_to_file",
     "audio_to_float32",
+    "audio_to_int16",
     "get_hf_turn_credentials",
     "get_twilio_turn_credentials",
     "get_turn_credentials",

backend/fastrtc/reply_on_pause.py

@@ -63,7 +63,7 @@ class AppState:
 
 ReplyFnGenerator = (
     Callable[
-        [tuple[int, NDArray[np.int16]], list[dict[Any, Any]]],
+        [tuple[int, NDArray[np.int16]], Any],
         Generator[EmitType, None, None],
     ]
     | Callable[
@@ -75,7 +75,7 @@ ReplyFnGenerator = (
         AsyncGenerator[EmitType, None],
     ]
     | Callable[
-        [tuple[int, NDArray[np.int16]], list[dict[Any, Any]]],
+        [tuple[int, NDArray[np.int16]], Any],
         AsyncGenerator[EmitType, None],
     ]
 )

backend/fastrtc/stream.py

@@ -62,6 +62,7 @@ class Stream(WebRTCConnectionMixin):
         additional_outputs: list[Component] | None = None,
         ui_args: UIArgs | None = None,
     ):
+        WebRTCConnectionMixin.__init__(self)
         self.mode = mode
         self.modality = modality
         self.rtp_params = rtp_params

backend/fastrtc/utils.py

@@ -294,6 +294,40 @@ def audio_to_float32(
     return audio[1].astype(np.float32) / 32768.0
 
 
+def audio_to_int16(
+    audio: tuple[int, NDArray[np.int16 | np.float32]],
+) -> NDArray[np.int16]:
+    """
+    Convert an audio tuple containing sample rate and numpy array data to int16.
+
+    Parameters
+    ----------
+    audio : tuple[int, np.ndarray]
+        A tuple containing:
+            - sample_rate (int): The audio sample rate in Hz
+            - data (np.ndarray): The audio data as a numpy array
+
+    Returns
+    -------
+    np.ndarray
+        The audio data as a numpy array with dtype int16
+
+    Example
+    -------
+    >>> sample_rate = 44100
+    >>> audio_data = np.array([0.1, -0.2, 0.3], dtype=np.float32)  # Example audio samples
+    >>> audio_tuple = (sample_rate, audio_data)
+    >>> audio_int16 = audio_to_int16(audio_tuple)
+    """
+    if audio[1].dtype == np.int16:
+        return audio[1]
+    elif audio[1].dtype == np.float32:
+        # Convert float32 to int16 by scaling to the int16 range
+        return (audio[1] * 32767.0).astype(np.int16)
+    else:
+        raise TypeError(f"Unsupported audio data type: {audio[1].dtype}")
+
+
 def aggregate_bytes_to_16bit(chunks_iterator):
     """
     Aggregate bytes to 16-bit audio samples.

backend/fastrtc/webrtc.py

@@ -122,6 +122,7 @@ class WebRTC(Component, WebRTCConnectionMixin):
             button_labels: Text to display on the audio or video start, stop, waiting buttons. Dict with keys "start", "stop", "waiting" mapping to the text to display on the buttons.
             icon_radius: Border radius of the icon button expressed as a percentage of the button size. Default is 50%
         """
+        WebRTCConnectionMixin.__init__(self)
         self.time_limit = time_limit
         self.height = height
         self.width = width
@@ -230,15 +231,9 @@ class WebRTC(Component, WebRTCConnectionMixin):
             inputs = [inputs]
             inputs = list(inputs)
 
-        def handler(webrtc_id: str, *args):
+        async def handler(webrtc_id: str, *args):
-            if self.additional_outputs[webrtc_id].queue.qsize() > 0:
+            async for next_outputs in self.output_stream(webrtc_id):
-                next_outputs = self.additional_outputs[webrtc_id].queue.get_nowait()
+                yield fn(*args, *next_outputs.args)  # type: ignore
-                return fn(*args, *next_outputs.args)  # type: ignore
-            return (
-                tuple([None for _ in range(len(outputs))])
-                if isinstance(outputs, Iterable)
-                else None
-            )
 
         return self.state_change(  # type: ignore
             fn=handler,
@@ -247,9 +242,9 @@ class WebRTC(Component, WebRTCConnectionMixin):
             js=js,
             concurrency_limit=concurrency_limit,
             concurrency_id=concurrency_id,
-            show_progress=show_progress,
+            show_progress="minimal",
             queue=queue,
-            trigger_mode="multiple",
+            trigger_mode="once",
         )
 
     def stream(

backend/fastrtc/webrtc_connection_mixin.py

@@ -35,7 +35,6 @@ from fastrtc.tracks import (
 )
 from fastrtc.utils import (
     AdditionalOutputs,
-    DataChannel,
     create_message,
     webrtc_error_handler,
 )
@@ -64,18 +63,20 @@ class OutputQueue:
 
 
 class WebRTCConnectionMixin:
-    pcs: set[RTCPeerConnection] = set([])
+    def __init__(self):
-    relay = MediaRelay()
+        self.pcs = set([])
-    connections: dict[str, list[Track]] = defaultdict(list)
+        self.relay = MediaRelay()
-    data_channels: dict[str, DataChannel] = {}
+        self.connections = defaultdict(list)
-    additional_outputs: dict[str, OutputQueue] = defaultdict(OutputQueue)
+        self.data_channels = {}
-    handlers: dict[str, HandlerType | Callable] = {}
+        self.additional_outputs = defaultdict(OutputQueue)
-    connection_timeouts: dict[str, asyncio.Event] = defaultdict(asyncio.Event)
+        self.handlers = {}
-    concurrency_limit: int | float
+        self.connection_timeouts = defaultdict(asyncio.Event)
-    event_handler: HandlerType
+        # These attributes should be set by subclasses:
-    time_limit: float | int | None
+        self.concurrency_limit: int | float | None
-    modality: Literal["video", "audio", "audio-video"]
+        self.event_handler: HandlerType | None
-    mode: Literal["send", "receive", "send-receive"]
+        self.time_limit: float | None
+        self.modality: Literal["video", "audio", "audio-video"]
+        self.mode: Literal["send", "receive", "send-receive"]
 
     @staticmethod
     async def wait_for_time_limit(pc: RTCPeerConnection, time_limit: float):

demo/phonic_chat/app.py

@@ -1,7 +1,3 @@
-import subprocess
-
-subprocess.run(["pip", "install", "fastrtc==0.0.4.post1"])
-
 import asyncio
 import base64
 import os
@@ -80,12 +76,11 @@ class PhonicHandler(AsyncStreamHandler):
         return super().shutdown()
 
 
-def add_to_chatbot(state, chatbot, message):
+def add_to_chatbot(chatbot, message):
-    state.append(message)
+    chatbot.append(message)
-    return state, gr.skip()
+    return chatbot
 
 
-state = gr.State(value=[])
 chatbot = gr.Chatbot(type="messages", value=[])
 stream = Stream(
     handler=PhonicHandler(),
@@ -99,7 +94,7 @@ stream = Stream(
             info="Select a voice from the dropdown",
         )
     ],
-    additional_outputs=[state, chatbot],
+    additional_outputs=[chatbot],
     additional_outputs_handler=add_to_chatbot,
     ui_args={
         "title": "Phonic Chat (Powered by FastRTC ⚡️)",
@@ -109,8 +104,8 @@ stream = Stream(
     time_limit=90 if get_space() else None,
 )
 
-with stream.ui:
+# with stream.ui:
-    state.change(lambda s: s, inputs=state, outputs=chatbot)
+#     state.change(lambda s: s, inputs=state, outputs=chatbot)
 
 if __name__ == "__main__":
     if (mode := os.getenv("MODE")) == "UI":

demo/talk_to_sambanova/app.py

@@ -38,6 +38,7 @@ def response(
 ):
     gradio_chatbot = gradio_chatbot or []
     conversation_state = conversation_state or []
+    print("chatbot", gradio_chatbot)
 
     text = stt_model.stt(audio)
     sample_rate, array = audio
@@ -47,7 +48,6 @@ def response(
     yield AdditionalOutputs(gradio_chatbot, conversation_state)
 
     conversation_state.append({"role": "user", "content": text})
-
     request = client.chat.completions.create(
         model="meta-llama/Llama-3.2-3B-Instruct",
         messages=conversation_state,  # type: ignore

docs/userguide/streams.md

@@ -10,7 +10,7 @@ from fastrtc import Stream
 import gradio as gr
 import numpy as np
 
-def detection(image):
+def detection(image, slider):
     return np.flip(image, axis=0)
 
 stream = Stream(

frontend/Index.svelte

@@ -148,6 +148,7 @@
 			{icon}
 			{icon_button_color}
 			{pulse_color}
+			{icon_radius}
 			{button_labels}
 			on:clear={() => gradio.dispatch("clear")}
 			on:play={() => gradio.dispatch("play")}

frontend/shared/InteractiveVideo.svelte

@@ -30,6 +30,7 @@
 	export let icon: string | undefined | ComponentType = undefined;
 	export let icon_button_color: string = "var(--color-accent)";
 	export let pulse_color: string = "var(--color-accent)";
+	export let icon_radius: number = 50;
 
 	const dispatch = createEventDispatcher<{
 		change: FileData | null;
@@ -62,8 +63,8 @@
 		{icon}
 		{icon_button_color}
 		{pulse_color}
-		{button_labels}
 		{icon_radius}
+		{button_labels}
 		on:error
 		on:start_recording
 		on:stop_recording

frontend/shared/Webcam.svelte

@@ -33,6 +33,7 @@
 	export let rtp_params: RTCRtpParameters = {} as RTCRtpParameters;
 	export let icon: string | undefined | ComponentType = undefined;
 	export let icon_button_color: string = "var(--color-accent)";
+	export let icon_radius: number = 50;
 	export let pulse_color: string = "var(--color-accent)";
 	export let button_labels: { start: string; stop: string; waiting: string };
 
@@ -242,6 +243,7 @@
 				icon={icon || Mic}
 				{icon_button_color}
 				{pulse_color}
+				{icon_radius}
 			/>
 		</div>
 	{/if}

pyproject.toml

@@ -8,7 +8,7 @@ build-backend = "hatchling.build"
 
 [project]
 name = "fastrtc"
-version = "0.0.8post1"
+version = "0.0.9"
 description = "The realtime communication library for Python"
 readme = "README.md"
 license = "apache-2.0"