moonlight-stream/moonlight-common-c
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Adjust initial audio resync drop time based on connection handshake latency

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This commit is contained in:
Cameron Gutman 2021-04-26 21:39:36 -05:00
parent 7174caf5f1
commit 33c4e98152
1 changed files with 17 additions and 1 deletions
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18
src/AudioStream.c
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@ -18,6 +18,7 @@ static PPLT_CRYPTO_CONTEXT audioDecryptionCtx;
static unsigned short lastSeq;
static bool receivedDataFromPeer;
static uint64_t pingStartTime;
#define RTP_PORT 48000
@ -76,6 +77,15 @@ int initializeAudioStream(void) {
return LastSocketFail();
}
// Track the time we started pinging. Audio samples will begin arriving
// shortly after the first ping reaches the host. However, we won't be
// ready to start playback until later on. As a result, we'll drop samples
// equivalent to the amount of time before the receive thread is ready
// to accept traffic in real-time.
// FIXME: This could also take into account the size of the recv buffer,
// since long RTSP handshakes could cause that to fill to capacity.
pingStartTime = PltGetMillis();
// We may receive audio before our threads are started, but that's okay. We'll
// drop the first 1 second of audio packets to catch up with the backlog.
int err = PltCreateThread("AudioPing", UdpPingThreadProc, NULL, &udpPingThread);
@ -185,9 +195,15 @@ static void ReceiveThreadProc(void* context) {
PQUEUED_AUDIO_PACKET packet;
int queueStatus;
bool useSelect;
int packetsToDrop = 1000 / AudioPacketDuration;
int initialResyncDelay = PltGetMillis() - pingStartTime;
int packetsToDrop;
int waitingForAudioMs;
// Cap delay at 3 seconds to account for recv buffer cap
initialResyncDelay = initialResyncDelay > 3000 ? 3000 : initialResyncDelay;
packetsToDrop = initialResyncDelay / AudioPacketDuration;
Limelog("Initial audio resync period: %d milliseconds\n", initialResyncDelay);
packet = NULL;
if (setNonFatalRecvTimeoutMs(rtpSocket, UDP_RECV_POLL_TIMEOUT_MS) < 0) {