bool HTTP4CLIProtocol::SignalInputData(IOBuffer &buffer) {
//1. Get the HTTP protocol. We are sure is a PT_INBOUND_HTTP
//because we return true inside AllowFarProtocol only when type == PT_INBOUND_HTTP
InboundHTTPProtocol *pHTTP = (InboundHTTPProtocol *) GetFarProtocol();
//2. Get the request headers
Variant headers = pHTTP->GetHeaders();
//3. Populate the input buffer for the next protocol in the stack (PT_INBOUND_JSONCLI)
//with the data we just found out inside the headers
URI uri;
string dummy = "http://localhost" + (string) headers[HTTP_FIRST_LINE][HTTP_URL];
FINEST("dummy: %s",STR(dummy));
if (!URI::FromString(dummy, false, uri)) {
FATAL("Invalid request");
return false;
}
string fullCommand=uri.document;
fullCommand+=" ";
if(uri.parameters.size()!=0){
fullCommand+=unb64(MAP_VAL(uri.parameters.begin()));
}
fullCommand+="\n";
_localInputBuffer.ReadFromString(fullCommand);
//4. Call the next protocol with the new buffer
return GetNearProtocol()->SignalInputData(_localInputBuffer);
}
bool InboundConnectivity::Initialize() {
//1. get the application
BaseClientApplication *pApplication = _pRTSP->GetApplication();
if (pApplication == NULL) {
FATAL("RTSP protocol not yet assigned to an application");
return false;
}
//2. Compute the bandwidthHint
uint32_t bandwidth = 0;
if (_videoTrack != V_NULL) {
bandwidth += (uint32_t) SDP_TRACK_BANDWIDTH(_videoTrack);
}
if (_audioTrack != V_NULL) {
bandwidth += (uint32_t) SDP_TRACK_BANDWIDTH(_audioTrack);
}
if (bandwidth == 0) {
bandwidth = _bandwidthHint;
}
//5. Create the in stream
if (_streamName == "")
_streamName = format("rtsp_%u", _pRTSP->GetId());
if (!pApplication->StreamNameAvailable(_streamName, _pRTSP)) {
FATAL("Stream name %s already taken", STR(_streamName));
return false;
}
_pInStream = new InNetRTPStream(_pRTSP, pApplication->GetStreamsManager(),
_streamName,
_videoTrack != V_NULL ? unb64((string) SDP_VIDEO_CODEC_H264_SPS(_videoTrack)) : "",
_videoTrack != V_NULL ? unb64((string) SDP_VIDEO_CODEC_H264_PPS(_videoTrack)) : "",
_audioTrack != V_NULL ? unhex(SDP_AUDIO_CODEC_SETUP(_audioTrack)) : "",
bandwidth,
_rtcpDetectionInterval);
//6. override the width/height with the values in session (if any)
Variant &session = _pRTSP->GetCustomParameters();
if ((session.HasKeyChain(_V_NUMERIC, true, 3, "customParameters", "externalStreamConfig", "width"))
&& (session.HasKeyChain(_V_NUMERIC, true, 3, "customParameters", "externalStreamConfig", "height"))) {
StreamCapabilities *pCap = _pInStream->GetCapabilities();
if (pCap->videoCodecId == CODEC_VIDEO_AVC) {
pCap->avc._widthOverride = (uint32_t) session["customParameters"]["externalStreamConfig"]["width"];
pCap->avc._heightOverride = (uint32_t) session["customParameters"]["externalStreamConfig"]["height"];
}
}
// pCap->avc._widthOverride=session["width"];
// pCap->avc._widthOverride=session[""];
//6. make the stream known to inbound RTP protocols
//and plug in the connectivity
if (_pRTPVideo != NULL) {
_pRTPVideo->SetStream(_pInStream, false);
_pRTPVideo->SetInbboundConnectivity(this);
_pRTCPVideo->SetInbboundConnectivity(this, false);
}
if (_pRTPAudio != NULL) {
_pRTPAudio->SetStream(_pInStream, true);
_pRTPAudio->SetInbboundConnectivity(this);
_pRTCPAudio->SetInbboundConnectivity(this, true);
}
//7. Pickup all outbound waiting streams
map<uint32_t, BaseOutStream *> subscribedOutStreams =
pApplication->GetStreamsManager()->GetWaitingSubscribers(
_streamName, _pInStream->GetType(), true);
//FINEST("subscribedOutStreams count: %"PRIz"u", subscribedOutStreams.size());
//8. Bind the waiting subscribers
FOR_MAP(subscribedOutStreams, uint32_t, BaseOutStream *, i) {
BaseOutStream *pBaseOutStream = MAP_VAL(i);
pBaseOutStream->Link(_pInStream);
}
void CommonTestsSuite::test_unb64() {
TS_ASSERT(unb64("dGhpcyBpcyBhIHRlc3Qx") == "this is a test1");
TS_ASSERT(unb64("dGhpcyBpcyBhIHRlc3Q=") == "this is a test");
TS_ASSERT(unb64("dGhpcyBpcyBhIHRlcw==") == "this is a tes");
TS_ASSERT(unb64("") == "");
}
string unb64(string source) {
return unb64((uint8_t *) STR(source), source.length());
}
