moonlight-stream/moonlight-common-c
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Dynamically determine audio, video, and control ports from RTSP DESCRIBE response

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This commit is contained in:
Cameron Gutman
2021-07-01 23:45:33 -05:00
parent 2660a05084
commit 56ccd99cc7
6 changed files with 101 additions and 16 deletions
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67
src/RtspConnection.c
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@@ -228,7 +228,7 @@ static bool transactRtspMessageTcp(PRTSP_MESSAGE request, PRTSP_MESSAGE response
// returns HTTP 200 OK for the /launch request before the RTSP handshake port
// is listening.
do {
sock = connectTcpSocket(&RemoteAddr, RemoteAddrLen, 48010, RTSP_TIMEOUT_SEC);
sock = connectTcpSocket(&RemoteAddr, RemoteAddrLen, RtspPortNumber, RTSP_TIMEOUT_SEC);
if (sock == INVALID_SOCKET) {
*error = LastSocketError();
if (*error == ECONNREFUSED) {
@@ -525,6 +525,57 @@ static int parseOpusConfigFromParamString(char* paramStr, int channelCount, POPU
return 0;
}
static bool parseMediaEntry(PRTSP_MESSAGE response, const char* mediaType, const char* transport, uint16_t* port) {
char paramsPrefix[128];
char paramsSuffix[128];
char* paramStart;
sprintf(paramsPrefix, "m=%s ", mediaType);
sprintf(paramsSuffix, " %s", transport);
// Look for the next match
paramStart = response->payload;
while ((paramStart = strstr(paramStart, paramsPrefix)) != NULL) {
// Skip the prefix
paramStart += strlen(paramsPrefix);
// The first part should be the port number
char* nextToken;
long int rawPort = strtol(paramStart, &nextToken, 10);
if (rawPort <= 0 || rawPort >= 65535) {
continue;
}
// Skip this entry if the transport isn't what we expect
if (strncmp(nextToken, paramsSuffix, strlen(paramsSuffix)) != 0) {
continue;
}
// This entry is a match
*port = (uint16_t)rawPort;
return true;
}
// No match for this media type and transport
return false;
}
static void parsePortConfigurations(PRTSP_MESSAGE response) {
// Don't parse ports on very old GFE versions
if (!APP_VERSION_AT_LEAST(7, 0, 0)) {
return;
}
parseMediaEntry(response, "video", "RTP/AVP", &VideoPortNumber);
parseMediaEntry(response, "audio", "RTP/AVP", &AudioPortNumber);
if (!parseMediaEntry(response, "application", "udp", &ControlPortNumber)) {
if (!parseMediaEntry(response, "application", "udp_enc", &ControlPortNumber)) {
parseMediaEntry(response, "application", "udp_ag", &ControlPortNumber);
}
}
}
// Parses the Opus configuration from an RTSP DESCRIBE response
static int parseOpusConfigurations(PRTSP_MESSAGE response) {
HighQualitySurroundSupported = false;
@@ -644,7 +695,7 @@ int performRtspHandshake(void) {
// Initialize global state
useEnet = (AppVersionQuad[0] >= 5) && (AppVersionQuad[0] <= 7) && (AppVersionQuad[2] < 404);
sprintf(rtspTargetUrl, "rtsp%s://%s:48010", useEnet ? "ru" : "", urlAddr);
sprintf(rtspTargetUrl, "rtsp%s://%s:%u", useEnet ? "ru" : "", urlAddr, RtspPortNumber);
currentSeqNumber = 1;
hasSessionId = false;
controlStreamId = APP_VERSION_AT_LEAST(7, 1, 431) ? "streamid=control/13/0" : "streamid=control/1/0";
@@ -676,7 +727,7 @@ int performRtspHandshake(void) {
ENetEvent event;
enet_address_set_address(&address, (struct sockaddr *)&RemoteAddr, RemoteAddrLen);
enet_address_set_port(&address, 48010);
enet_address_set_port(&address, RtspPortNumber);
// Create a client that can use 1 outgoing connection and 1 channel
client = enet_host_create(RemoteAddr.ss_family, NULL, 1, 1, 0, 0);
@@ -695,7 +746,7 @@ int performRtspHandshake(void) {
// Wait for the connect to complete
if (serviceEnetHost(client, &event, RTSP_TIMEOUT_SEC * 1000) <= 0 ||
event.type != ENET_EVENT_TYPE_CONNECT) {
Limelog("RTSP: Failed to connect to UDP port 48010\n");
Limelog("RTSP: Failed to connect to UDP port %u\n", RtspPortNumber);
enet_peer_reset(peer);
peer = NULL;
enet_host_destroy(client);
@@ -773,6 +824,14 @@ int performRtspHandshake(void) {
}
}
// Parse audio, video, and control ports out of the RTSP DESCRIBE response.
parsePortConfigurations(&response);
// Let the audio stream know the port number is now finalized.
// NB: This is needed because audio stream init happens before RTSP,
// which is not the case for the video stream.
notifyAudioPortNegotiationComplete();
// Parse the Opus surround parameters out of the RTSP DESCRIBE response.
ret = parseOpusConfigurations(&response);
if (ret != 0) {