#include "Limelight-internal.h"
#include "PlatformSockets.h"
#include "PlatformThreads.h"
#include "LinkedBlockingQueue.h"
#include "RtpReorderQueue.h"

static SOCKET rtpSocket = INVALID_SOCKET;

static LINKED_BLOCKING_QUEUE packetQueue;
static RTP_REORDER_QUEUE rtpReorderQueue;

static PLT_THREAD udpPingThread;
static PLT_THREAD receiveThread;
static PLT_THREAD decoderThread;

static unsigned short lastSeq;

#define RTP_PORT 48000

#define MAX_PACKET_SIZE 100
#define RTP_RECV_BUFFER 12 * MAX_PACKET_SIZE

typedef struct _QUEUED_AUDIO_PACKET {
	// data must remain at the front
	char data[MAX_PACKET_SIZE];

	int size;
	union {
		RTP_QUEUE_ENTRY rentry;
		LINKED_BLOCKING_QUEUE_ENTRY lentry;
	} q;
} QUEUED_AUDIO_PACKET, *PQUEUED_AUDIO_PACKET;

/* Initialize the audio stream */
void initializeAudioStream(void) {
	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		LbqInitializeLinkedBlockingQueue(&packetQueue, 30);
	}
	RtpqInitializeQueue(&rtpReorderQueue, RTPQ_DEFAULT_MAX_SIZE, RTPQ_DEFUALT_QUEUE_TIME);
    lastSeq = 0;
}

static void freePacketList(PLINKED_BLOCKING_QUEUE_ENTRY entry) {
	PLINKED_BLOCKING_QUEUE_ENTRY nextEntry;

	while (entry != NULL) {
		nextEntry = entry->flink;

		// The entry is stored within the data allocation
		free(entry->data);

		entry = nextEntry;
	}
}

/* Tear down the audio stream once we're done with it */
void destroyAudioStream(void) {
	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		freePacketList(LbqDestroyLinkedBlockingQueue(&packetQueue));
	}
	RtpqCleanupQueue(&rtpReorderQueue);
}

static void UdpPingThreadProc(void *context) {
	/* Ping in ASCII */
	char pingData[] = { 0x50, 0x49, 0x4E, 0x47 };
	struct sockaddr_in6 saddr;
	SOCK_RET err;

    memcpy(&saddr, &RemoteAddr, sizeof(saddr));
	saddr.sin6_port = htons(RTP_PORT);

	/* Send PING every 500 milliseconds */
	while (!PltIsThreadInterrupted(&udpPingThread)) {
		err = sendto(rtpSocket, pingData, sizeof(pingData), 0, (struct sockaddr*)&saddr, RemoteAddrLen);
		if (err != sizeof(pingData)) {
			Limelog("Audio Ping: sendto() failed: %d\n", (int)LastSocketError());
			ListenerCallbacks.connectionTerminated(LastSocketError());
			return;
		}

		PltSleepMs(500);
	}
}

static int queuePacketToLbq(PQUEUED_AUDIO_PACKET *packet) {
	int err;

	err = LbqOfferQueueItem(&packetQueue, *packet, &(*packet)->q.lentry);
	if (err == LBQ_SUCCESS) {
		// The LBQ owns the buffer now
		*packet = NULL;
	}
	else if (err == LBQ_BOUND_EXCEEDED) {
		Limelog("Audio packet queue overflow\n");
		freePacketList(LbqFlushQueueItems(&packetQueue));
	}
	else if (err == LBQ_INTERRUPTED) {
		free(*packet);
		return 0;
	}

	return 1;
}

static void decodeInputData(PQUEUED_AUDIO_PACKET packet) {
	PRTP_PACKET rtp;

	rtp = (PRTP_PACKET) &packet->data[0];
	if (lastSeq != 0 && (unsigned short) (lastSeq + 1) != rtp->sequenceNumber) {
		Limelog("Received OOS audio data (expected %d, but got %d)\n", lastSeq + 1, rtp->sequenceNumber);

		AudioCallbacks.decodeAndPlaySample(NULL, 0);
	}

	lastSeq = rtp->sequenceNumber;

	AudioCallbacks.decodeAndPlaySample((char *) (rtp + 1), packet->size - sizeof(*rtp));
}

static void ReceiveThreadProc(void* context) {
	PRTP_PACKET rtp;
	PQUEUED_AUDIO_PACKET packet;
	int queueStatus;

	packet = NULL;

	while (!PltIsThreadInterrupted(&receiveThread)) {
		if (packet == NULL) {
			packet = (PQUEUED_AUDIO_PACKET) malloc(sizeof(*packet));
			if (packet == NULL) {
				Limelog("Audio Receive: malloc() failed\n");
				ListenerCallbacks.connectionTerminated(-1);
				return;
			}
		}

		packet->size = (int) recv(rtpSocket, &packet->data[0], MAX_PACKET_SIZE, 0);
		if (packet->size <= 0) {
			Limelog("Audio Receive: recv() failed: %d\n", (int)LastSocketError());
			free(packet);
			ListenerCallbacks.connectionTerminated(LastSocketError());
			return;
		}

		if (packet->size < sizeof(RTP_PACKET)) {
			// Runt packet
			continue;
		}

		rtp = (PRTP_PACKET) &packet->data[0];
		if (rtp->packetType != 97) {
			// Not audio
			continue;
		}
        
        // RTP sequence number must be in host order for the RTP queue
        rtp->sequenceNumber = htons(rtp->sequenceNumber);

		queueStatus = RtpqAddPacket(&rtpReorderQueue, (PRTP_PACKET) packet, &packet->q.rentry);
		if (queueStatus == RTPQ_RET_HANDLE_IMMEDIATELY) {
			if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
				if (!queuePacketToLbq(&packet)) {
					// An exit signal was received
					return;
				}
			} else {
				decodeInputData(packet);
			}
		}
		else {
			if (queueStatus != RTPQ_RET_REJECTED) {
				// The queue consumed our packet, so we must allocate a new one
				packet = NULL;
			}

			if (queueStatus == RTPQ_RET_QUEUED_PACKETS_READY) {
				// If packets are ready, pull them and send them to the decoder
				while ((packet = (PQUEUED_AUDIO_PACKET) RtpqGetQueuedPacket(&rtpReorderQueue)) != NULL) {
					if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
						if (!queuePacketToLbq(&packet)) {
							// An exit signal was received
							return;
						}
					} else {
						decodeInputData(packet);
					}
				}
			}
		}
	}
}

static void DecoderThreadProc(void* context) {
	int err;
	PQUEUED_AUDIO_PACKET packet;

	while (!PltIsThreadInterrupted(&decoderThread)) {
		err = LbqWaitForQueueElement(&packetQueue, (void**) &packet);
		if (err != LBQ_SUCCESS) {
            // An exit signal was received
			return;
		}

		decodeInputData(packet);

		free(packet);
	}
}

void stopAudioStream(void) {
	PltInterruptThread(&udpPingThread);
	PltInterruptThread(&receiveThread);
	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		PltInterruptThread(&decoderThread);
	}

	if (rtpSocket != INVALID_SOCKET) {
		closesocket(rtpSocket);
		rtpSocket = INVALID_SOCKET;
	}

	PltJoinThread(&udpPingThread);
	PltJoinThread(&receiveThread);
	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		PltJoinThread(&decoderThread);
	}

	PltCloseThread(&udpPingThread);
	PltCloseThread(&receiveThread);
	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		PltCloseThread(&decoderThread);
	}

    AudioCallbacks.cleanup();
}

int startAudioStream(void) {
	int err;
    
    AudioCallbacks.init();

	rtpSocket = bindUdpSocket(RemoteAddr.ss_family, RTP_RECV_BUFFER);
	if (rtpSocket == INVALID_SOCKET) {
		return LastSocketFail();
	}

	err = PltCreateThread(UdpPingThreadProc, NULL, &udpPingThread);
	if (err != 0) {
		return err;
	}

	err = PltCreateThread(ReceiveThreadProc, NULL, &receiveThread);
	if (err != 0) {
		return err;
	}

	if ((AudioCallbacks.capabilities & CAPABILITY_DIRECT_SUBMIT) == 0) {
		err = PltCreateThread(DecoderThreadProc, NULL, &decoderThread);
		if (err != 0) {
			return err;
		}
	}

	return 0;
}