implementation

backend/gradio_webrtc/utils.py

@@ -0,0 +1,63 @@
+import time
+import fractions
+import av
+import asyncio
+import threading
+from typing import Callable
+
+AUDIO_PTIME = 0.020
+
+
+def player_worker_decode(
+    loop,
+    callable: Callable,
+    stream,
+    queue: asyncio.Queue,
+    throttle_playback: bool,
+    thread_quit: threading.Event,
+):
+    audio_sample_rate = 48000
+    audio_samples = 0
+    audio_time_base = fractions.Fraction(1, audio_sample_rate)
+    audio_resampler = av.AudioResampler(
+        format="s16",
+        layout="stereo",
+        rate=audio_sample_rate,
+        frame_size=int(audio_sample_rate * AUDIO_PTIME),
+    )
+
+    frame_time = None
+    start_time = time.time()
+    generator = None
+
+    while not thread_quit.is_set():
+        print("stream.latest_args", stream.latest_args)
+        if stream.latest_args == "not_set":
+            continue
+        if generator is None:
+            generator = callable(*stream.latest_args)
+        try:
+            frame = next(generator)
+        except Exception as exc:
+            if isinstance(exc, StopIteration):
+                print("Not iterating")
+                asyncio.run_coroutine_threadsafe(queue.put(frame), loop)
+                thread_quit.set()
+            break
+
+        # read up to 1 second ahead
+        if throttle_playback:
+            elapsed_time = time.time() - start_time
+            if frame_time and frame_time > elapsed_time + 1:
+                time.sleep(0.1)
+        sample_rate, audio_array = frame
+        frame = av.AudioFrame.from_ndarray(audio_array, format="s16", layout="mono")
+        frame.sample_rate = sample_rate
+        for frame in audio_resampler.resample(frame):
+            # fix timestamps
+            frame.pts = audio_samples
+            frame.time_base = audio_time_base
+            audio_samples += frame.samples
+
+            frame_time = frame.time
+            asyncio.run_coroutine_threadsafe(queue.put(frame), loop)