fix timestamp may overflow

webrtc.py

@@ -29,7 +29,7 @@ import numpy as np
 
 AUDIO_PTIME = 0.020  # 20ms audio packetization
 VIDEO_CLOCK_RATE = 90000
-VIDEO_PTIME = 1 / 25  # 30fps
+VIDEO_PTIME = 0.040 #1 / 25  # 30fps
 VIDEO_TIME_BASE = fractions.Fraction(1, VIDEO_CLOCK_RATE)
 SAMPLE_RATE = 16000
 AUDIO_TIME_BASE = fractions.Fraction(1, SAMPLE_RATE)
@@ -56,6 +56,7 @@ class PlayerStreamTrack(MediaStreamTrack):
         self._player = player
         self._queue = asyncio.Queue()
         self.timelist = [] #记录最近包的时间戳
+        self.current_frame_count = 0
         if self.kind == 'video':
             self.framecount = 0
             self.lasttime = time.perf_counter()
@@ -72,7 +73,8 @@ class PlayerStreamTrack(MediaStreamTrack):
             if hasattr(self, "_timestamp"):
                 #self._timestamp = (time.time()-self._start) * VIDEO_CLOCK_RATE
                 self._timestamp += int(VIDEO_PTIME * VIDEO_CLOCK_RATE)
-                wait = self._start + (self._timestamp / VIDEO_CLOCK_RATE) - time.time()
+                self.current_frame_count += 1
+                wait = self._start + self.current_frame_count * VIDEO_PTIME - time.time()
                 # wait = self.timelist[0] + len(self.timelist)*VIDEO_PTIME - time.time()               
                 if wait>0:
                     await asyncio.sleep(wait)
@@ -89,7 +91,8 @@ class PlayerStreamTrack(MediaStreamTrack):
             if hasattr(self, "_timestamp"):
                 #self._timestamp = (time.time()-self._start) * SAMPLE_RATE
                 self._timestamp += int(AUDIO_PTIME * SAMPLE_RATE)
-                wait = self._start + (self._timestamp / SAMPLE_RATE) - time.time()
+                self.current_frame_count += 1
+                wait = self._start + self.current_frame_count * AUDIO_PTIME - time.time()
                 # wait = self.timelist[0] + len(self.timelist)*AUDIO_PTIME - time.time()
                 if wait>0:
                     await asyncio.sleep(wait)