Tune SO_RCVBUF logic

- Calculate desired SO_RCVBUF by packet size for the video socket
- Use the OS default SO_RCVBUF for audio and STUN sockets

src/AudioStream.c

@@ -25,11 +25,6 @@ static uint8_t opusHeaderByte;
 
 #define MAX_PACKET_SIZE 1400
 
-// This is much larger than we should typically have buffered, but
-// it needs to be. We need a cushion in case our thread gets blocked
-// for longer than normal.
-#define RTP_RECV_BUFFER (64 * 1024)
-
 typedef struct _QUEUE_AUDIO_PACKET_HEADER {
     LINKED_BLOCKING_QUEUE_ENTRY lentry;
     int size;
@@ -88,7 +83,7 @@ int initializeAudioStream(void) {
 
     // For GFE 3.22 compatibility, we must start the audio ping thread before the RTSP handshake.
     // It will not reply to our RTSP PLAY request until the audio ping has been received.
-    rtpSocket = bindUdpSocket(RemoteAddr.ss_family, RTP_RECV_BUFFER);
+    rtpSocket = bindUdpSocket(RemoteAddr.ss_family, 0);
     if (rtpSocket == INVALID_SOCKET) {
         return LastSocketFail();
     }

src/PlatformSockets.c

@@ -266,6 +266,7 @@ SOCKET bindUdpSocket(int addrfamily, int bufferSize) {
     }
 #endif
 
+    if (bufferSize != 0) {
         // We start at the requested recv buffer value and step down until we find
         // a value that the OS will accept.
         for (;;) {
@@ -295,7 +296,15 @@ SOCKET bindUdpSocket(int addrfamily, int bufferSize) {
         else {
             Limelog("Unable to set receive buffer size: %d\n", LastSocketError());
         }
+
+        {
+            SOCKADDR_LEN len = sizeof(bufferSize);
+            if (getsockopt(s, SOL_SOCKET, SO_RCVBUF, (char*)&bufferSize, &len) == 0) {
+                Limelog("Actual receive buffer size: %d\n", bufferSize);
+            }
+        }
 #endif
+    }
 
     return s;
 }

src/SimpleStun.c

@@ -73,7 +73,7 @@ int LiFindExternalAddressIP4(const char* stunServer, unsigned short stunPort, un
         goto Exit;
     }
 
-    sock = bindUdpSocket(hints.ai_family, 2048);
+    sock = bindUdpSocket(hints.ai_family, 0);
     if (sock == INVALID_SOCKET) {
         err = LastSocketFail();
         Limelog("Failed to connect to STUN server: %d\n", err);

src/VideoStream.c

@@ -5,8 +5,6 @@
 
 #define FIRST_FRAME_PORT 47996
 
-#define RTP_RECV_BUFFER (512 * 1024)
-
 static RTP_VIDEO_QUEUE rtpQueue;
 
 static SOCKET rtpSocket = INVALID_SOCKET;
@@ -25,6 +23,14 @@ static bool receivedFullFrame;
 // the RTP queue will wait for missing/reordered packets.
 #define RTP_QUEUE_DELAY 10
 
+// This is the desired number of video packets that can be
+// stored in the socket's receive buffer. 2048 is chosen
+// because it should be large enough for all reasonable
+// frame sizes (probably 2 or 3 frames) without using too
+// much kernel memory with larger packet sizes. It also
+// can smooth over transient pauses in network traffic
+// and subsequent packet/frame bursts that follow.
+#define RTP_RECV_PACKETS_BUFFERED 2048
 
 // Initialize the video stream
 void initializeVideoStream(void) {
@@ -254,7 +260,7 @@ int startVideoStream(void* rendererContext, int drFlags) {
         return err;
     }
 
-    rtpSocket = bindUdpSocket(RemoteAddr.ss_family, RTP_RECV_BUFFER);
+    rtpSocket = bindUdpSocket(RemoteAddr.ss_family, RTP_RECV_PACKETS_BUFFERED * (StreamConfig.packetSize + MAX_RTP_HEADER_SIZE));
     if (rtpSocket == INVALID_SOCKET) {
         VideoCallbacks.cleanup();
         return LastSocketError();