moonlight-stream/moonlight-common-c
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Implement audio FEC recovery support

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This commit is contained in:
Cameron Gutman
2021-06-01 18:31:56 -05:00
parent 122ce4a568
commit 89918324ce
6 changed files with 486 additions and 322 deletions
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52
src/AudioStream.c
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@@ -3,7 +3,7 @@
static SOCKET rtpSocket = INVALID_SOCKET;
static LINKED_BLOCKING_QUEUE packetQueue;
static RTP_REORDER_QUEUE rtpReorderQueue;
static RTP_AUDIO_QUEUE rtpAudioQueue;
static PLT_THREAD udpPingThread;
static PLT_THREAD receiveThread;
@@ -26,15 +26,14 @@ static uint64_t firstReceiveTime;
// for longer than normal.
#define RTP_RECV_BUFFER (64 * 1024)
typedef struct _QUEUED_AUDIO_PACKET {
// data must remain at the front
char data[MAX_PACKET_SIZE];
typedef struct _QUEUE_AUDIO_PACKET_HEADER {
LINKED_BLOCKING_QUEUE_ENTRY lentry;
int size;
union {
RTP_QUEUE_ENTRY rentry;
LINKED_BLOCKING_QUEUE_ENTRY lentry;
} q;
} QUEUED_AUDIO_PACKET_HEADER, *PQUEUED_AUDIO_PACKET_HEADER;
typedef struct _QUEUED_AUDIO_PACKET {
QUEUED_AUDIO_PACKET_HEADER header;
char data[MAX_PACKET_SIZE];
} QUEUED_AUDIO_PACKET, *PQUEUED_AUDIO_PACKET;
static void UdpPingThreadProc(void* context) {
@@ -67,7 +66,7 @@ static void UdpPingThreadProc(void* context) {
// Initialize the audio stream and start
int initializeAudioStream(void) {
LbqInitializeLinkedBlockingQueue(&packetQueue, 30);
RtpqInitializeQueue(&rtpReorderQueue, RTPQ_DEFAULT_MAX_SIZE, RTPQ_DEFAULT_QUEUE_TIME);
RtpaInitializeQueue(&rtpAudioQueue);
lastSeq = 0;
receivedDataFromPeer = false;
firstReceiveTime = 0;
@@ -122,13 +121,13 @@ void destroyAudioStream(void) {
PltDestroyCryptoContext(audioDecryptionCtx);
freePacketList(LbqDestroyLinkedBlockingQueue(&packetQueue));
RtpqCleanupQueue(&rtpReorderQueue);
RtpaCleanupQueue(&rtpAudioQueue);
}
static bool queuePacketToLbq(PQUEUED_AUDIO_PACKET* packet) {
int err;
err = LbqOfferQueueItem(&packetQueue, *packet, &(*packet)->q.lentry);
err = LbqOfferQueueItem(&packetQueue, *packet, &(*packet)->header.lentry);
if (err == LBQ_SUCCESS) {
// The LBQ owns the buffer now
*packet = NULL;
@@ -160,7 +159,7 @@ static void decodeInputData(PQUEUED_AUDIO_PACKET packet) {
// We must have room for the AES padding which may be written to the buffer
unsigned char decryptedOpusData[ROUND_TO_PKCS7_PADDED_LEN(MAX_PACKET_SIZE)];
unsigned char iv[16] = { 0 };
int dataLength = packet->size - sizeof(*rtp);
int dataLength = packet->header.size - sizeof(*rtp);
LC_ASSERT(dataLength <= MAX_PACKET_SIZE);
@@ -182,7 +181,7 @@ static void decodeInputData(PQUEUED_AUDIO_PACKET packet) {
AudioCallbacks.decodeAndPlaySample((char*)decryptedOpusData, dataLength);
}
else {
AudioCallbacks.decodeAndPlaySample((char*)(rtp + 1), packet->size - sizeof(*rtp));
AudioCallbacks.decodeAndPlaySample((char*)(rtp + 1), packet->header.size - sizeof(*rtp));
}
}
@@ -217,13 +216,13 @@ static void ReceiveThreadProc(void* context) {
}
}
packet->size = recvUdpSocket(rtpSocket, &packet->data[0], MAX_PACKET_SIZE, useSelect);
if (packet->size < 0) {
packet->header.size = recvUdpSocket(rtpSocket, &packet->data[0], MAX_PACKET_SIZE, useSelect);
if (packet->header.size < 0) {
Limelog("Audio Receive: recvUdpSocket() failed: %d\n", (int)LastSocketError());
ListenerCallbacks.connectionTerminated(LastSocketFail());
break;
}
else if (packet->size == 0) {
else if (packet->header.size == 0) {
// Receive timed out; try again
if (!receivedDataFromPeer) {
@@ -236,16 +235,12 @@ static void ReceiveThreadProc(void* context) {
continue;
}
if (packet->size < (int)sizeof(RTP_PACKET)) {
if (packet->header.size < (int)sizeof(RTP_PACKET)) {
// Runt packet
continue;
}
rtp = (PRTP_PACKET)&packet->data[0];
if (rtp->packetType != 97) {
// Not audio
continue;
}
if (!receivedDataFromPeer) {
receivedDataFromPeer = true;
@@ -260,7 +255,10 @@ static void ReceiveThreadProc(void* context) {
// GFE accumulates audio samples before we are ready to receive them, so
// we will drop the ones that arrived before the receive thread was ready.
if (packetsToDrop > 0) {
packetsToDrop--;
// Only count actual audio data (not FEC) in the packets to drop calculation
if (rtp->packetType == 97) {
packetsToDrop--;
}
continue;
}
@@ -269,7 +267,7 @@ static void ReceiveThreadProc(void* context) {
rtp->timestamp = BE32(rtp->timestamp);
rtp->ssrc = BE32(rtp->ssrc);
queueStatus = RtpqAddPacket(&rtpReorderQueue, (PRTP_PACKET)packet, &packet->q.rentry);
queueStatus = RtpaAddPacket(&rtpAudioQueue, (PRTP_PACKET)&packet->data[0], (uint16_t)packet->header.size);
if (RTPQ_HANDLE_NOW(queueStatus)) {
if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
if (!queuePacketToLbq(&packet)) {
@@ -289,7 +287,11 @@ static void ReceiveThreadProc(void* context) {
if (RTPQ_PACKET_READY(queueStatus)) {
// If packets are ready, pull them and send them to the decoder
while ((packet = (PQUEUED_AUDIO_PACKET)RtpqGetQueuedPacket(&rtpReorderQueue)) != NULL) {
uint16_t length;
while ((packet = (PQUEUED_AUDIO_PACKET)RtpaGetQueuedPacket(&rtpAudioQueue, sizeof(QUEUED_AUDIO_PACKET_HEADER), &length)) != NULL) {
// Populate header data (not preserved in queued packets)
packet->header.size = length;
if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
if (!queuePacketToLbq(&packet)) {
// An exit signal was received