static int
rtmp_probe(const char *url0, char *errbuf, size_t errlen, int timeout_ms)
{
RTMP *r;
char *url = mystrdupa(url0);
r = RTMP_Alloc();
RTMP_Init(r, NULL);
if(!RTMP_SetupURL(r, url)) {
snprintf(errbuf, errlen, "Unable to setup RTMP-session");
RTMP_Free(r);
return BACKEND_PROBE_FAIL;
}
if(!RTMP_Connect(r, NULL, errbuf, errlen, timeout_ms)) {
RTMP_Close(r);
RTMP_Free(r);
return BACKEND_PROBE_FAIL;
}
RTMP_SetReadTimeout(r, timeout_ms);
if(!RTMP_ConnectStream(r, 0)) {
snprintf(errbuf, errlen, "Unable to connect RTMP-stream");
RTMP_Close(r);
RTMP_Free(r);
return BACKEND_PROBE_FAIL;
}
RTMP_Close(r);
RTMP_Free(r);
return BACKEND_PROBE_OK;
}
/*
* Class: net_butterflytv_rtmp_client_RtmpClient
* Method: open
* Signature: (Ljava/lang/String;)I
*/
JNIEXPORT jint JNICALL Java_net_ossrs_sea_RtmpClient_open
(JNIEnv * env, jobject thiz, jstring url_, jboolean isPublishMode) {
const char *url = (*env)->GetStringUTFChars(env, url_, 0);
rtmp = RTMP_Alloc();
if (rtmp == NULL) {
return -1;
}
RTMP_Init(rtmp);
int ret = RTMP_SetupURL(rtmp, url);
if (!ret) {
RTMP_Free(rtmp);
return -2;
}
if (isPublishMode) {
RTMP_EnableWrite(rtmp);
}
ret = RTMP_Connect(rtmp, NULL);
if (!ret) {
RTMP_Free(rtmp);
return -3;
}
ret = RTMP_ConnectStream(rtmp, 0);
if (!ret) {
return -4;
}
(*env)->ReleaseStringUTFChars(env, url_, url);
return 1;
}
static gboolean
gst_rtmp_sink_start (GstBaseSink * basesink)
{
GstRTMPSink *sink = GST_RTMP_SINK (basesink);
if (!sink->uri) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
("Please set URI for RTMP output"), ("No URI set before starting"));
return FALSE;
}
sink->rtmp_uri = g_strdup (sink->uri);
sink->rtmp = RTMP_Alloc ();
RTMP_Init (sink->rtmp);
if (!RTMP_SetupURL (sink->rtmp, sink->rtmp_uri)) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Failed to setup URL '%s'", sink->uri));
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
return FALSE;
}
GST_DEBUG_OBJECT (sink, "Created RTMP object");
/* Mark this as an output connection */
RTMP_EnableWrite (sink->rtmp);
sink->first = TRUE;
return TRUE;
}
void RTMPWriter::close() {
if(rtmp) {
RTMP_Close(rtmp);
RTMP_Free(rtmp);
rtmp = NULL;
}
}
/* open the file, do stuff necessary to go to PAUSED state */
static gboolean
gst_rtmp_src_start (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
gchar *uri_copy;
src = GST_RTMP_SRC (basesrc);
if (!src->uri) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
return FALSE;
}
src->cur_offset = 0;
src->last_timestamp = 0;
src->discont = TRUE;
uri_copy = g_strdup (src->uri);
src->rtmp = RTMP_Alloc ();
RTMP_Init (src->rtmp);
if (!RTMP_SetupURL (src->rtmp, uri_copy)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Failed to setup URL '%s'", src->uri));
g_free (uri_copy);
RTMP_Free (src->rtmp);
src->rtmp = NULL;
return FALSE;
}
src->seekable = !(src->rtmp->Link.lFlags & RTMP_LF_LIVE);
GST_INFO_OBJECT (src, "seekable %d", src->seekable);
/* open if required */
if (!RTMP_IsConnected (src->rtmp)) {
if (!RTMP_Connect (src->rtmp, NULL)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Could not connect to RTMP stream \"%s\" for reading", src->uri));
RTMP_Free (src->rtmp);
src->rtmp = NULL;
return FALSE;
}
}
return TRUE;
}
static GstFlowReturn
gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstRTMPSink *sink = GST_RTMP_SINK (bsink);
GstBuffer *reffed_buf = NULL;
if (sink->first) {
/* open the connection */
if (!RTMP_IsConnected (sink->rtmp)) {
if (!RTMP_Connect (sink->rtmp, NULL)
|| !RTMP_ConnectStream (sink->rtmp, 0)) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Could not connect to RTMP stream \"%s\" for writing", sink->uri));
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (sink, "Opened connection to %s", sink->rtmp_uri);
}
/* FIXME: Parse the first buffer and see if it contains a header plus a packet instead
* of just assuming it's only the header */
GST_LOG_OBJECT (sink, "Caching first buffer of size %d for concatenation",
GST_BUFFER_SIZE (buf));
gst_buffer_replace (&sink->cache, buf);
sink->first = FALSE;
return GST_FLOW_OK;
}
if (sink->cache) {
GST_LOG_OBJECT (sink, "Joining 2nd buffer of size %d to cached buf",
GST_BUFFER_SIZE (buf));
gst_buffer_ref (buf);
reffed_buf = buf = gst_buffer_join (sink->cache, buf);
sink->cache = NULL;
}
GST_LOG_OBJECT (sink, "Sending %d bytes to RTMP server",
GST_BUFFER_SIZE (buf));
if (!RTMP_Write (sink->rtmp,
(char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf))) {
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_ERROR;
}
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_OK;
}
static CURLcode rtmp_disconnect(struct connectdata *conn,
bool dead_connection)
{
RTMP *r = conn->proto.generic;
(void)dead_connection;
if(r) {
conn->proto.generic = NULL;
RTMP_Close(r);
RTMP_Free(r);
}
return CURLE_OK;
}
static void
rtmp_free(rtmp_t *r)
{
if(r->vcodec != NULL)
media_codec_deref(r->vcodec);
if(r->acodec != NULL)
media_codec_deref(r->acodec);
RTMP_Close(r->r);
RTMP_Free(r->r);
}
int main(int argc, char **argv)
{
RTMP *rtmp=RTMP_Alloc();
if(!rtmp)
return 1;
RTMP_Init(rtmp);
RTMP_Free(rtmp);
return 0;
}
int rtmp_rvod_stop(t_rtmp_vod_ctx* ctx)
{
if (!ctx || !ctx->rtmp)
{
return -1;
}
RTMP *rtmp = ctx->rtmp;
rtmp_rvod_closefile(ctx);
RTMP_Close(rtmp);
RTMP_Free(rtmp);
return 0;
}
LibRtmp::~LibRtmp()
{
Close();
RTMP_Free(rtmp_);
if (streming_url_)
{
free(streming_url_);
streming_url_ = NULL;
}
if (flog_) fclose(flog_);
}
static CURLcode rtmp_setup_connection(struct connectdata *conn)
{
RTMP *r = RTMP_Alloc();
if(!r)
return CURLE_OUT_OF_MEMORY;
RTMP_Init(r);
RTMP_SetBufferMS(r, DEF_BUFTIME);
if(!RTMP_SetupURL(r, conn->data->change.url)) {
RTMP_Free(r);
return CURLE_URL_MALFORMAT;
}
conn->proto.generic = r;
return CURLE_OK;
}
RTMPWriter::~RTMPWriter() {
cb_disconnect = NULL;
cb_user = NULL;
if(rtmp && state == RW_STATE_INITIALIZED) {
RTMP_Close(rtmp);
}
if(rtmp) {
RTMP_Free(rtmp);
rtmp = NULL;
}
state = RW_STATE_NONE;
}
void RtmpSessionFini(RTMP_SESSION *pSession)
{
if(NULL != pSession->prtmp)
{
RTMP_Free(pSession->prtmp);
pSession->prtmp = NULL;
}
if(NULL != pSession->pPkt)
{
RTMPPacket_Free(pSession->pPkt);
free(pSession->pPkt);
pSession->pPkt = NULL;
}
return ;
}
static gboolean
gst_rtmp_src_stop (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (basesrc);
if (src->rtmp) {
RTMP_Free (src->rtmp);
src->rtmp = NULL;
}
src->cur_offset = 0;
src->last_timestamp = 0;
src->discont = TRUE;
return TRUE;
}
void sc_streamer_reconnect(sc_streamer *streamer) {
printf("Reconnecting \n");
RTMP_Close(streamer->rtmp);
RTMP_Free(streamer->rtmp);
streamer->rtmp = open_RTMP_stream( streamer->stream_uri );
streamer->reconnect_tries++;
if(streamer->reconnect_tries >= SC_Reconnect_Attempts) {
printf("Reconnected failed after %i tries \n", streamer->reconnect_tries);
if(streamer->so_name != NULL) {
sc_streamer_stop_cursor(streamer);
} else {
sc_streamer_stop_video(streamer);
}
exit(1);
}
}
static gboolean
gst_rtmp_sink_stop (GstBaseSink * basesink)
{
GstRTMPSink *sink = GST_RTMP_SINK (basesink);
gst_buffer_replace (&sink->cache, NULL);
if (sink->rtmp) {
RTMP_Close (sink->rtmp);
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
}
if (sink->rtmp_uri) {
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
}
return TRUE;
}
void RTMPWriter::write(uint8_t* data, size_t nbytes) {
if(state == RW_STATE_NONE) {
printf("error: cannot write to rtmp server because we haven't been initialized. did you call initialize()?\n");
return;
}
else if(state == RW_STATE_RECONNECTING) {
//printf("reconnecting... ignoring data...\n");
return;
}
else if(state == RW_STATE_DISCONNECTED) {
// the caller needs to call reconnect()
return;
}
//printf("rtmp: %ld\n", nbytes);
int r = RTMP_Write(rtmp, (const char*)data, (int)nbytes);
if(r < 0) {
// @todo - we should close and cleanup here!!!!
RTMP_Close(rtmp);
RTMP_Free(rtmp);
rtmp = NULL;
printf("error: something went wrong while trying to write data to the rtmp server.\n");
if(state == RW_STATE_DISCONNECTED) {
return;
}
// when initialized and we arrive here, it means we're disconnected
else if(state == RW_STATE_INITIALIZED) {
state = RW_STATE_DISCONNECTED;
if(cb_disconnect) {
cb_disconnect(this, cb_user);
}
}
}
}
RTMPPublisher::~RTMPPublisher()
{
bStopping = true;
//we're in the middle of connecting! wait for that to happen to avoid all manner of race conditions
if (hConnectionThread)
{
WaitForSingleObject(hConnectionThread, INFINITE);
OSCloseThread(hConnectionThread);
}
if(hSendThread)
{
ReleaseSemaphore(hSendSempahore, 1, NULL);
//wake it up in case it's waiting for buffer space
SetEvent(hBufferSpaceAvailableEvent);
OSTerminateThread(hSendThread, 20000);
}
if(hSendSempahore)
CloseHandle(hSendSempahore);
if(hDataMutex)
OSCloseMutex(hDataMutex);
while (bufferedPackets.Num()) {
bufferedPackets[0].data.Clear();
bufferedPackets.Remove(0);
}
//wake up and shut down the buffered sender
SetEvent(hWriteEvent);
SetEvent(hBufferEvent);
if (hSocketThread)
{
OSTerminateThread(hSocketThread, 20000);
//at this point nothing new should be coming in to the buffer, flush out what remains
FlushDataBuffer();
}
if(rtmp)
{
//disable the buffered send, so RTMP_Close writes directly to the net
rtmp->m_bCustomSend = 0;
RTMP_Close(rtmp);
}
if (dataBuffer)
Free(dataBuffer);
if (hDataBufferMutex)
OSCloseMutex(hDataBufferMutex);
if (hBufferEvent)
CloseHandle(hBufferEvent);
if (hBufferSpaceAvailableEvent)
CloseHandle(hBufferSpaceAvailableEvent);
if (hWriteEvent)
CloseHandle(hWriteEvent);
if(rtmp)
RTMP_Free(rtmp);
//--------------------------
for(UINT i=0; i<queuedPackets.Num(); i++)
queuedPackets[i].data.Clear();
queuedPackets.Clear();
double dBFrameDropPercentage = double(numBFramesDumped)/NumTotalVideoFrames()*100.0;
double dPFrameDropPercentage = double(numPFramesDumped)/NumTotalVideoFrames()*100.0;
Log(TEXT("Number of times waited to send: %d, Waited for a total of %d bytes"), totalTimesWaited, totalBytesWaited);
Log(TEXT("Number of b-frames dropped: %u (%0.2g%%), Number of p-frames dropped: %u (%0.2g%%), Total %u (%0.2g%%)"),
numBFramesDumped, dBFrameDropPercentage,
numPFramesDumped, dPFrameDropPercentage,
numBFramesDumped+numPFramesDumped, dBFrameDropPercentage+dPFrameDropPercentage);
/*if(totalCalls)
Log(TEXT("average send time: %u"), totalTime/totalCalls);*/
strRTMPErrors.Clear();
//--------------------------
}
DWORD WINAPI RTMPPublisher::CreateConnectionThread(RTMPPublisher *publisher)
{
//------------------------------------------------------
// set up URL
bool bRetry = false;
bool bSuccess = false;
bool bCanRetry = false;
String failReason;
String strBindIP;
int serviceID = AppConfig->GetInt (TEXT("Publish"), TEXT("Service"));
String strURL = AppConfig->GetString(TEXT("Publish"), TEXT("URL"));
String strPlayPath = AppConfig->GetString(TEXT("Publish"), TEXT("PlayPath"));
strURL.KillSpaces();
strPlayPath.KillSpaces();
LPSTR lpAnsiURL = NULL, lpAnsiPlaypath = NULL;
RTMP *rtmp = NULL;
//--------------------------------
// unbelievably disgusting hack for elgato devices
String strOldDirectory;
UINT dirSize = GetCurrentDirectory(0, 0);
strOldDirectory.SetLength(dirSize);
GetCurrentDirectory(dirSize, strOldDirectory.Array());
OSSetCurrentDirectory(API->GetAppPath());
//--------------------------------
if(!strURL.IsValid())
{
failReason = TEXT("No server specified to connect to");
goto end;
}
if(serviceID != 0)
{
XConfig serverData;
if(!serverData.Open(TEXT("services.xconfig")))
{
failReason = TEXT("Could not open services.xconfig");
goto end;
}
XElement *services = serverData.GetElement(TEXT("services"));
if(!services)
{
failReason = TEXT("Could not find any services in services.xconfig");
goto end;
}
XElement *service = NULL;
DWORD numServices = services->NumElements();
for(UINT i=0; i<numServices; i++)
{
XElement *curService = services->GetElementByID(i);
if(curService->GetInt(TEXT("id")) == serviceID)
{
service = curService;
break;
}
}
if(!service)
{
failReason = TEXT("Could not find the service specified in services.xconfig");
goto end;
}
XElement *servers = service->GetElement(TEXT("servers"));
if(!servers)
{
failReason = TEXT("Could not find any servers for the service specified in services.xconfig");
goto end;
}
XDataItem *item = servers->GetDataItem(strURL);
if(!item)
item = servers->GetDataItemByID(0);
strURL = item->GetData();
Log(TEXT("Using RTMP service: %s"), service->GetName());
Log(TEXT(" Server selection: %s"), strURL.Array());
}
//------------------------------------------------------
// now back to the elgato directory if it needs the directory changed still to function *sigh*
OSSetCurrentDirectory(strOldDirectory);
//------------------------------------------------------
rtmp = RTMP_Alloc();
RTMP_Init(rtmp);
RTMP_LogSetCallback(librtmpErrorCallback);
//RTMP_LogSetLevel(RTMP_LOGERROR);
lpAnsiURL = strURL.CreateUTF8String();
lpAnsiPlaypath = strPlayPath.CreateUTF8String();
if(!RTMP_SetupURL2(rtmp, lpAnsiURL, lpAnsiPlaypath))
{
failReason = Str("Connection.CouldNotParseURL");
goto end;
}
char *rtmpUser = AppConfig->GetString(TEXT("Publish"), TEXT("Username")).CreateUTF8String();
char *rtmpPass = AppConfig->GetString(TEXT("Publish"), TEXT("Password")).CreateUTF8String();
if (rtmpUser)
{
rtmp->Link.pubUser.av_val = rtmpUser;
rtmp->Link.pubUser.av_len = (int)strlen(rtmpUser);
}
if (rtmpPass)
{
rtmp->Link.pubPasswd.av_val = rtmpPass;
rtmp->Link.pubPasswd.av_len = (int)strlen(rtmpPass);
}
RTMP_EnableWrite(rtmp); //set it to publish
rtmp->Link.swfUrl.av_len = rtmp->Link.tcUrl.av_len;
rtmp->Link.swfUrl.av_val = rtmp->Link.tcUrl.av_val;
/*rtmp->Link.pageUrl.av_len = rtmp->Link.tcUrl.av_len;
rtmp->Link.pageUrl.av_val = rtmp->Link.tcUrl.av_val;*/
rtmp->Link.flashVer.av_val = "FMLE/3.0 (compatible; FMSc/1.0)";
rtmp->Link.flashVer.av_len = (int)strlen(rtmp->Link.flashVer.av_val);
//-----------------------------------------
UINT tcpBufferSize = AppConfig->GetInt(TEXT("Publish"), TEXT("TCPBufferSize"), 64*1024);
if(tcpBufferSize < 8192)
tcpBufferSize = 8192;
else if(tcpBufferSize > 1024*1024)
tcpBufferSize = 1024*1024;
rtmp->m_outChunkSize = 4096;//RTMP_DEFAULT_CHUNKSIZE;//
rtmp->m_bSendChunkSizeInfo = TRUE;
rtmp->m_bUseNagle = TRUE;
strBindIP = AppConfig->GetString(TEXT("Publish"), TEXT("BindToIP"), TEXT("Default"));
if (scmp(strBindIP, TEXT("Default")))
{
rtmp->m_bindIP.addr.sin_family = AF_INET;
rtmp->m_bindIP.addrLen = sizeof(rtmp->m_bindIP.addr);
if (WSAStringToAddress(strBindIP.Array(), AF_INET, NULL, (LPSOCKADDR)&rtmp->m_bindIP.addr, &rtmp->m_bindIP.addrLen) == SOCKET_ERROR)
{
// no localization since this should rarely/never happen
failReason = TEXT("WSAStringToAddress: Could not parse address");
goto end;
}
}
LogInterfaceType(rtmp);
//-----------------------------------------
if(!RTMP_Connect(rtmp, NULL))
{
failReason = Str("Connection.CouldNotConnect");
failReason << TEXT("\r\n\r\n") << RTMPPublisher::GetRTMPErrors();
bCanRetry = true;
goto end;
}
if(!RTMP_ConnectStream(rtmp, 0))
{
failReason = Str("Connection.InvalidStream");
failReason << TEXT("\r\n\r\n") << RTMPPublisher::GetRTMPErrors();
bCanRetry = true;
goto end;
}
//-----------------------------------------
OSDebugOut(TEXT("Connected: %u\r\n"), OSGetTime());
publisher->RequestKeyframe(1000);
//-----------------------------------------
bSuccess = true;
end:
if (lpAnsiURL)
Free(lpAnsiURL);
if (lpAnsiPlaypath)
Free(lpAnsiPlaypath);
if(!bSuccess)
{
if(rtmp)
{
RTMP_Close(rtmp);
RTMP_Free(rtmp);
}
if(failReason.IsValid())
App->SetStreamReport(failReason);
if(!publisher->bStopping)
PostMessage(hwndMain, OBS_REQUESTSTOP, bCanRetry ? 0 : 1, 0);
Log(TEXT("Connection to %s failed: %s"), strURL.Array(), failReason.Array());
publisher->bStopping = true;
}
else
{
publisher->Init(rtmp, tcpBufferSize);
publisher->bConnected = true;
publisher->bConnecting = false;
}
return 0;
}
RTMPPublisher::~RTMPPublisher()
{
//OSDebugOut (TEXT("*** ~RTMPPublisher (%d queued, %d buffered, %d data)\n"), queuedPackets.Num(), bufferedPackets.Num(), curDataBufferLen);
bStopping = true;
//we're in the middle of connecting! wait for that to happen to avoid all manner of race conditions
if (hConnectionThread)
{
WaitForSingleObject(hConnectionThread, INFINITE);
OSCloseThread(hConnectionThread);
}
//send all remaining buffered packets, this may block since it respects timestamps
FlushBufferedPackets ();
//OSDebugOut (TEXT("%d queued after flush\n"), queuedPackets.Num());
if(hSendThread)
{
//this marks the thread to exit after current work is done
SetEvent(hSendLoopExit);
//these wake up the thread
ReleaseSemaphore(hSendSempahore, 1, NULL);
SetEvent(hBufferSpaceAvailableEvent);
//wait 50 sec for all data to finish sending
if (WaitForSingleObject(hSendThread, 50000) == WAIT_TIMEOUT)
{
Log(TEXT("~RTMPPublisher: Network appears stalled with %d / %d buffered, dropping connection!"), curDataBufferLen, dataBufferSize);
FatalSocketShutdown();
//this will wake up and flush the sendloop if it's still trying to send out stuff
ReleaseSemaphore(hSendSempahore, 1, NULL);
SetEvent(hBufferSpaceAvailableEvent);
}
OSTerminateThread(hSendThread, 10000);
}
if(hSendSempahore)
CloseHandle(hSendSempahore);
//OSDebugOut (TEXT("*** ~RTMPPublisher hSendThread terminated (%d queued, %d buffered, %d data)\n"), queuedPackets.Num(), bufferedPackets.Num(), curDataBufferLen);
if (hSocketThread)
{
//mark the socket loop to shut down after the buffer is empty
SetEvent(hSocketLoopExit);
//wake it up in case it already is empty
SetEvent(hBufferEvent);
//wait 60 sec for it to exit
OSTerminateThread(hSocketThread, 60000);
}
//OSDebugOut (TEXT("*** ~RTMPPublisher hSocketThread terminated (%d queued, %d buffered, %d data)\n"), queuedPackets.Num(), bufferedPackets.Num(), curDataBufferLen);
if(rtmp)
{
if (RTMP_IsConnected(rtmp))
{
//at this point nothing should be in the buffer, flush out what remains to the net and make it blocking
FlushDataBuffer();
//disable the buffered send, so RTMP_* functions write directly to the net (and thus block)
rtmp->m_bCustomSend = 0;
//manually shut down the stream and issue a graceful socket shutdown
RTMP_DeleteStream(rtmp);
shutdown(rtmp->m_sb.sb_socket, SD_SEND);
//this waits for the socket shutdown to complete gracefully
for (;;)
{
char buff[1024];
int ret;
ret = recv(rtmp->m_sb.sb_socket, buff, sizeof(buff), 0);
if (!ret)
break;
else if (ret == -1)
{
Log(TEXT("~RTMPublisher: Received error %d while waiting for graceful shutdown."), WSAGetLastError());
break;
}
}
//OSDebugOut(TEXT("Graceful shutdown complete.\n"));
}
//this closes the socket if not already done
RTMP_Close(rtmp);
}
if(hDataMutex)
OSCloseMutex(hDataMutex);
while (bufferedPackets.Num())
{
//this should not happen any more...
bufferedPackets[0].data.Clear();
bufferedPackets.Remove(0);
}
if (dataBuffer)
Free(dataBuffer);
if (hDataBufferMutex)
OSCloseMutex(hDataBufferMutex);
if (hBufferEvent)
CloseHandle(hBufferEvent);
if (hSendLoopExit)
CloseHandle(hSendLoopExit);
if (hSocketLoopExit)
CloseHandle(hSocketLoopExit);
if (hSendBacklogEvent)
CloseHandle(hSendBacklogEvent);
if (hBufferSpaceAvailableEvent)
CloseHandle(hBufferSpaceAvailableEvent);
if (hWriteEvent)
CloseHandle(hWriteEvent);
if(rtmp)
{
if (rtmp->Link.pubUser.av_val)
Free(rtmp->Link.pubUser.av_val);
if (rtmp->Link.pubPasswd.av_val)
Free(rtmp->Link.pubPasswd.av_val);
RTMP_Free(rtmp);
}
//--------------------------
for(UINT i=0; i<queuedPackets.Num(); i++)
queuedPackets[i].data.Clear();
queuedPackets.Clear();
double dBFrameDropPercentage = double(numBFramesDumped)/NumTotalVideoFrames()*100.0;
double dPFrameDropPercentage = double(numPFramesDumped)/NumTotalVideoFrames()*100.0;
if (totalSendCount)
Log(TEXT("Average send payload: %d bytes, average send interval: %d ms"), (DWORD)(totalSendBytes / totalSendCount), totalSendPeriod / totalSendCount);
Log(TEXT("Number of times waited to send: %d, Waited for a total of %d bytes"), totalTimesWaited, totalBytesWaited);
Log(TEXT("Number of b-frames dropped: %u (%0.2g%%), Number of p-frames dropped: %u (%0.2g%%), Total %u (%0.2g%%)"),
numBFramesDumped, dBFrameDropPercentage,
numPFramesDumped, dPFrameDropPercentage,
numBFramesDumped+numPFramesDumped, dBFrameDropPercentage+dPFrameDropPercentage);
/*if(totalCalls)
Log(TEXT("average send time: %u"), totalTime/totalCalls);*/
strRTMPErrors.Clear();
//--------------------------
}
RTMPPublisher::~RTMPPublisher()
{
//OSDebugOut (TEXT("*** ~RTMPPublisher (%d queued, %d buffered)\n"), queuedPackets.Num(), bufferedPackets.Num());
bStopping = true;
//we're in the middle of connecting! wait for that to happen to avoid all manner of race conditions
if (hConnectionThread)
{
WaitForSingleObject(hConnectionThread, INFINITE);
OSCloseThread(hConnectionThread);
}
FlushBufferedPackets ();
//OSDebugOut (TEXT("%d queued after flush\n"), queuedPackets.Num());
if(hSendThread)
{
//this marks the thread to exit after current work is done
SetEvent(hSendLoopExit);
//this wakes up the thread
ReleaseSemaphore(hSendSempahore, 1, NULL);
//wait 60 sec for it to exit
OSTerminateThread(hSendThread, 60000);
}
if(hSendSempahore)
CloseHandle(hSendSempahore);
//OSDebugOut (TEXT("*** ~RTMPPublisher hSendThread terminated (%d queued, %d buffered, %d data)\n"), queuedPackets.Num(), bufferedPackets.Num(), curDataBufferLen);
if (hSocketThread)
{
//mark the socket loop to shut down after the buffer is empty
SetEvent(hSocketLoopExit);
//wait 60 sec for it to exit
OSTerminateThread(hSocketThread, 60000);
}
//OSDebugOut (TEXT("*** ~RTMPPublisher hSocketThread terminated (%d queued, %d buffered, %d data)\n"), queuedPackets.Num(), bufferedPackets.Num(), curDataBufferLen);
if(rtmp)
{
//at this point nothing should be in the buffer, flush out what remains and make it blocking
FlushDataBuffer();
//disable the buffered send, so RTMP_Close writes directly to the net
rtmp->m_bCustomSend = 0;
//ideally we need some kind of delay here, since we just dumped several seconds worth of timestamps to the network
//at once, and Twitch at shows the offline screen as soon as the connection is severed even if there are
//pending video frames.
if (RTMP_IsConnected(rtmp))
Sleep (500); //for now
RTMP_Close(rtmp);
}
if(hDataMutex)
OSCloseMutex(hDataMutex);
while (bufferedPackets.Num())
{
//this should not happen any more...
bufferedPackets[0].data.Clear();
bufferedPackets.Remove(0);
}
if (dataBuffer)
Free(dataBuffer);
if (hDataBufferMutex)
OSCloseMutex(hDataBufferMutex);
if (hBufferEvent)
CloseHandle(hBufferEvent);
if (hSendLoopExit)
CloseHandle(hSendLoopExit);
if (hSocketLoopExit)
CloseHandle(hSocketLoopExit);
if (hSendBacklogEvent)
CloseHandle(hSendBacklogEvent);
if (hBufferSpaceAvailableEvent)
CloseHandle(hBufferSpaceAvailableEvent);
if (hWriteEvent)
CloseHandle(hWriteEvent);
if(rtmp)
RTMP_Free(rtmp);
//--------------------------
for(UINT i=0; i<queuedPackets.Num(); i++)
queuedPackets[i].data.Clear();
queuedPackets.Clear();
double dBFrameDropPercentage = double(numBFramesDumped)/NumTotalVideoFrames()*100.0;
double dPFrameDropPercentage = double(numPFramesDumped)/NumTotalVideoFrames()*100.0;
Log(TEXT("Number of times waited to send: %d, Waited for a total of %d bytes"), totalTimesWaited, totalBytesWaited);
Log(TEXT("Number of b-frames dropped: %u (%0.2g%%), Number of p-frames dropped: %u (%0.2g%%), Total %u (%0.2g%%)"),
numBFramesDumped, dBFrameDropPercentage,
numPFramesDumped, dPFrameDropPercentage,
numBFramesDumped+numPFramesDumped, dBFrameDropPercentage+dPFrameDropPercentage);
/*if(totalCalls)
Log(TEXT("average send time: %u"), totalTime/totalCalls);*/
strRTMPErrors.Clear();
//--------------------------
}
int main(int argc, char *argv[])
{
struct stat sb;
int opt, i;
int chapter = 0, available = 0;
char *book = NULL;
char audio[256];
while ((opt = getopt(argc, argv, "v:a:b:c:l")) != -1) {
switch (opt) {
case 'v':
version = optarg;
break;
case 'a':
author = optarg;
break;
case 'b':
book = optarg;
break;
case 'c':
chapter = atoi(optarg);
break;
case 'l':
return list();
}
}
sprintf(audio, "%s-%s", version, author);
for (i = 0; i < ARRAY_SIZE(audios); i++) {
if (strcasecmp(audios[i].audio, audio) == 0) {
available = 1;
break;
}
}
if (!available) {
fprintf(stderr, "Invalid version `%s' or author `%s'\n", version, author);
return EINVAL;
}
for (i = 0; i < ARRAY_SIZE(books); i++) {
if (!book || strcasecmp(books[i].osis, book) == 0) {
available = 1;
break;
}
}
if (!available) {
fprintf(stderr, "Invalid book `%s'\n", book);
return EINVAL;
}
if ((rtmp = RTMP_Alloc()) == NULL || (buffer = malloc(BUFSIZ)) == NULL) {
perror("malloc");
return ENOMEM;
}
if (stat(audio, &sb) == -1) {
if (errno == ENOENT) {
mkdir(audio, 0755);
} else {
perror("stat");
return errno;
}
} else if (!S_ISDIR(sb.st_mode)) {
fprintf(stderr, "%s is not a directory\n", audio);
return errno;
}
if (book) {
if (chapter > books[i].chapter) {
fprintf(stderr, "Invalid chapter `%d'\n", chapter);
return EINVAL;
}
if (chapter) {
download(books[i].osis, chapter);
} else {
for (chapter = 1; chapter <= books[i].chapter; chapter++) {
download(books[i].osis, chapter);
}
}
} else {
for (i = 0; i < ARRAY_SIZE(books); i++) {
for (chapter = 1; chapter <= books[i].chapter; chapter++) {
download(books[i].osis, chapter);
}
}
}
RTMP_Free(rtmp);
free(buffer);
return 0;
}
//Publish using RTMP_SendPacket()
int publish_using_packet(){
RTMP *rtmp=NULL;
RTMPPacket *packet=NULL;
uint32_t start_time=0;
uint32_t now_time=0;
//the timestamp of the previous frame
long pre_frame_time=0;
long lasttime=0;
int bNextIsKey=1;
uint32_t preTagsize=0;
//packet attributes
uint32_t type=0;
uint32_t datalength=0;
uint32_t timestamp=0;
uint32_t streamid=0;
FILE*fp=NULL;
fp=fopen("../live.flv","rb");
if (!fp){
RTMP_LogPrintf("Open File Error.\n");
// CleanupSockets();
return -1;
}
/* set log level */
//RTMP_LogLevel loglvl=RTMP_LOGDEBUG;
//RTMP_LogSetLevel(loglvl);
// if (!InitSockets()){
// RTMP_LogPrintf("Init Socket Err\n");
// return -1;
// }
rtmp=RTMP_Alloc();
RTMP_Init(rtmp);
//set connection timeout,default 30s
rtmp->Link.timeout=5;
if(!RTMP_SetupURL(rtmp,"rtmp://101.251.251.93:1935/myapp/mystream"))
{
RTMP_Log(RTMP_LOGERROR,"SetupURL Err\n");
RTMP_Free(rtmp);
// CleanupSockets();
return -1;
}
//if unable,the AMF command would be 'play' instead of 'publish'
RTMP_EnableWrite(rtmp);
if (!RTMP_Connect(rtmp,NULL)){
RTMP_Log(RTMP_LOGERROR,"Connect Err\n");
RTMP_Free(rtmp);
// CleanupSockets();
return -1;
}
if (!RTMP_ConnectStream(rtmp,0)){
RTMP_Log(RTMP_LOGERROR,"ConnectStream Err\n");
RTMP_Close(rtmp);
RTMP_Free(rtmp);
// CleanupSockets();
return -1;
}
packet=(RTMPPacket*)malloc(sizeof(RTMPPacket));
RTMPPacket_Alloc(packet,1024*64);
RTMPPacket_Reset(packet);
packet->m_hasAbsTimestamp = 0;
packet->m_nChannel = 0x04;
packet->m_nInfoField2 = rtmp->m_stream_id;
RTMP_LogPrintf("Start to send data ...\n");
//jump over FLV Header
fseek(fp,9,SEEK_SET);
//jump over previousTagSizen
fseek(fp,4,SEEK_CUR);
start_time=RTMP_GetTime();
while(1)
{
if((((now_time=RTMP_GetTime())-start_time)
<(pre_frame_time)) && bNextIsKey){
//wait for 1 sec if the send process is too fast
//this mechanism is not very good,need some improvement
if(pre_frame_time>lasttime){
RTMP_LogPrintf("TimeStamp:%8lu ms\n",pre_frame_time);
lasttime=pre_frame_time;
}
sleep(1);
continue;
}
//not quite the same as FLV spec
if(!ReadU8(&type,fp))
break;
if(!ReadU24(&datalength,fp))
break;
if(!ReadTime(×tamp,fp))
break;
if(!ReadU24(&streamid,fp))
break;
if (type!=0x08&&type!=0x09){
//jump over non_audio and non_video frame，
//jump over next previousTagSizen at the same time
fseek(fp,datalength+4,SEEK_CUR);
continue;
}
if(fread(packet->m_body,1,datalength,fp)!=datalength)
break;
packet->m_headerType = RTMP_PACKET_SIZE_LARGE;
packet->m_nTimeStamp = timestamp;
packet->m_packetType = type;
packet->m_nBodySize = datalength;
pre_frame_time=timestamp;
if (!RTMP_IsConnected(rtmp)){
RTMP_Log(RTMP_LOGERROR,"rtmp is not connect\n");
break;
}
if (!RTMP_SendPacket(rtmp,packet,0)){
RTMP_Log(RTMP_LOGERROR,"Send Error\n");
break;
}
if(!ReadU32(&preTagsize,fp))
break;
if(!PeekU8(&type,fp))
break;
if(type==0x09){
if(fseek(fp,11,SEEK_CUR)!=0)
break;
if(!PeekU8(&type,fp)){
break;
}
if(type==0x17)
bNextIsKey=1;
else
bNextIsKey=0;
fseek(fp,-11,SEEK_CUR);
}
}
RTMP_LogPrintf("\nSend Data Over\n");
if(fp)
fclose(fp);
if (rtmp!=NULL){
RTMP_Close(rtmp);
RTMP_Free(rtmp);
rtmp=NULL;
}
if (packet!=NULL){
RTMPPacket_Free(packet);
free(packet);
packet=NULL;
}
// CleanupSockets();
return 0;
}
//Publish using RTMP_Write()
int publish_using_write(){
uint32_t start_time=0;
uint32_t now_time=0;
uint32_t pre_frame_time=0;
uint32_t lasttime=0;
int bNextIsKey=0;
char* pFileBuf=NULL;
//read from tag header
uint32_t type=0;
uint32_t datalength=0;
uint32_t timestamp=0;
RTMP *rtmp=NULL;
FILE*fp=NULL;
fp=fopen("cuc_ieschool.flv","rb");
if (!fp){
RTMP_LogPrintf("Open File Error.\n");
// CleanupSockets();
return -1;
}
/* set log level */
//RTMP_LogLevel loglvl=RTMP_LOGDEBUG;
//RTMP_LogSetLevel(loglvl);
// if (!InitSockets()){
// RTMP_LogPrintf("Init Socket Err\n");
// return -1;
// }
rtmp=RTMP_Alloc();
RTMP_Init(rtmp);
//set connection timeout,default 30s
rtmp->Link.timeout=5;
if(!RTMP_SetupURL(rtmp,"rtmp://localhost/publishlive/livestream"))
{
RTMP_Log(RTMP_LOGERROR,"SetupURL Err\n");
RTMP_Free(rtmp);
// CleanupSockets();
return -1;
}
RTMP_EnableWrite(rtmp);
//1hour
RTMP_SetBufferMS(rtmp, 3600*1000);
if (!RTMP_Connect(rtmp,NULL)){
RTMP_Log(RTMP_LOGERROR,"Connect Err\n");
RTMP_Free(rtmp);
// CleanupSockets();
return -1;
}
if (!RTMP_ConnectStream(rtmp,0)){
RTMP_Log(RTMP_LOGERROR,"ConnectStream Err\n");
RTMP_Close(rtmp);
RTMP_Free(rtmp);
//CleanupSockets();
return -1;
}
printf("Start to send data ...\n");
//jump over FLV Header
fseek(fp,9,SEEK_SET);
//jump over previousTagSizen
fseek(fp,4,SEEK_CUR);
start_time=RTMP_GetTime();
while(1)
{
if((((now_time=RTMP_GetTime())-start_time)
<(pre_frame_time)) && bNextIsKey){
//wait for 1 sec if the send process is too fast
//this mechanism is not very good,need some improvement
if(pre_frame_time>lasttime){
RTMP_LogPrintf("TimeStamp:%8lu ms\n", pre_frame_time);
lasttime=pre_frame_time;
}
sleep(1);
continue;
}
//jump over type
fseek(fp,1,SEEK_CUR);
if(!ReadU24(&datalength,fp))
break;
if(!ReadTime(×tamp,fp))
break;
//jump back
fseek(fp,-8,SEEK_CUR);
pFileBuf=(char*)malloc(11+datalength+4);
memset(pFileBuf,0,11+datalength+4);
if(fread(pFileBuf,1,11+datalength+4,fp)!=(11+datalength+4))
break;
pre_frame_time=timestamp;
if (!RTMP_IsConnected(rtmp)){
RTMP_Log(RTMP_LOGERROR,"rtmp is not connect\n");
break;
}
if (!RTMP_Write(rtmp,pFileBuf,11+datalength+4)){
RTMP_Log(RTMP_LOGERROR,"Rtmp Write Error\n");
break;
}
free(pFileBuf);
pFileBuf=NULL;
if(!PeekU8(&type,fp))
break;
if(type==0x09){
if(fseek(fp,11,SEEK_CUR)!=0)
break;
if(!PeekU8(&type,fp)){
break;
}
if(type==0x17)
bNextIsKey=1;
else
bNextIsKey=0;
fseek(fp,-11,SEEK_CUR);
}
}
RTMP_LogPrintf("\nSend Data Over\n");
if(fp)
fclose(fp);
if (rtmp!=NULL){
RTMP_Close(rtmp);
RTMP_Free(rtmp);
rtmp=NULL;
}
if(pFileBuf){
free(pFileBuf);
pFileBuf=NULL;
}
//CleanupSockets();
return 0;
}
static GstFlowReturn
gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstRTMPSink *sink = GST_RTMP_SINK (bsink);
GstBuffer *reffed_buf = NULL;
GstMapInfo map;
if (sink->first) {
/* open the connection */
if (!RTMP_IsConnected (sink->rtmp)) {
if (!RTMP_Connect (sink->rtmp, NULL)
|| !RTMP_ConnectStream (sink->rtmp, 0)) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Could not connect to RTMP stream \"%s\" for writing", sink->uri));
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (sink, "Opened connection to %s", sink->rtmp_uri);
}
/* FIXME: Parse the first buffer and see if it contains a header plus a packet instead
* of just assuming it's only the header */
GST_LOG_OBJECT (sink, "Caching first buffer of size %" G_GSIZE_FORMAT
" for concatenation", gst_buffer_get_size (buf));
gst_buffer_replace (&sink->cache, buf);
sink->first = FALSE;
return GST_FLOW_OK;
}
if (sink->cache) {
GST_LOG_OBJECT (sink, "Joining 2nd buffer of size %" G_GSIZE_FORMAT
" to cached buf", gst_buffer_get_size (buf));
gst_buffer_ref (buf);
reffed_buf = buf = gst_buffer_append (sink->cache, buf);
sink->cache = NULL;
}
GST_LOG_OBJECT (sink, "Sending %" G_GSIZE_FORMAT " bytes to RTMP server",
gst_buffer_get_size (buf));
gst_buffer_map (buf, &map, GST_MAP_READ);
if (RTMP_Write (sink->rtmp, (char *) map.data, map.size) <= 0)
goto write_failed;
gst_buffer_unmap (buf, &map);
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_OK;
/* ERRORS */
write_failed:
{
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
gst_buffer_unmap (buf, &map);
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_ERROR;
}
}
/*
* Class: net_butterflytv_rtmp_client_RtmpClient
* Method: close
* Signature: ()I
*/
JNIEXPORT jint JNICALL Java_net_ossrs_sea_RtmpClient_close
(JNIEnv * env, jobject thiz) {
RTMP_Close(rtmp);
RTMP_Free(rtmp);
return 0;
}
int main(int argc, char* argv[])
{
InitSockets();
double duration = -1;
int nRead;
//is live stream ?
bool bLiveStream = TRUE;
int bufsize = 1024 * 1024 * 10;
char *buf = (char*)malloc(bufsize);
memset(buf, 0, bufsize);
long countbufsize = 0;
FILE *fp = fopen("receive.flv", "wb");
if (!fp)
{
RTMP_LogPrintf("Open File Error.\n");
CleanupSockets();
return -1;
}
/* set log level */
RTMP_LogLevel loglvl = RTMP_LOGDEBUG;
RTMP_LogSetLevel(loglvl);
RTMP *rtmp = RTMP_Alloc();
RTMP_Init(rtmp);
//set connection timeout,default 30s
rtmp->Link.timeout = 10;
char* rtmpFilePath = "d:\\rtmp.raw";
rtmp->m_pRTMPFile = fopen(rtmpFilePath,"rb");
if (!rtmp->m_pRTMPFile)
{
RTMP_LogPrintf("Failed to open File :%s\n", rtmpFilePath);
return FALSE;
}
// HKS's live URL
if(!RTMP_SetupURL(rtmp, "rtmp://live.hkstv.hk.lxdns.com/live/hks"))
{
RTMP_Log(RTMP_LOGERROR, "SetupURL Err\n");
RTMP_Free(rtmp);
CleanupSockets();
return -1;
}
if (bLiveStream)
{
rtmp->Link.lFlags |= RTMP_LF_LIVE;
}
//1hour
RTMP_SetBufferMS(rtmp, 3600 * 1000);
if(!RTMP_Connect(rtmp, NULL))
{
RTMP_Log(RTMP_LOGERROR, "Connect Err\n");
RTMP_Free(rtmp);
CleanupSockets();
return -1;
}
if(!RTMP_ConnectStream(rtmp, 0))
{
RTMP_Log(RTMP_LOGERROR, "ConnectStream Err\n");
RTMP_Close(rtmp);
RTMP_Free(rtmp);
CleanupSockets();
return -1;
}
while(nRead = RTMP_Read(rtmp, buf, bufsize))
{
fwrite(buf, 1, nRead, fp);
countbufsize += nRead;
RTMP_LogPrintf("Receive: %5dByte, Total: %5.2fkB\n", nRead, countbufsize * 1.0 / 1024);
}
if(fp)
fclose(fp);
if(fpPcap)
fclose(fpPcap);
if (rtmp->m_pRTMPFile)
{
fclose(rtmp->m_pRTMPFile);
}
if(buf)
{
free(buf);
}
if(rtmp)
{
RTMP_Close(rtmp);
RTMP_Free(rtmp);
CleanupSockets();
rtmp = NULL;
}
return 0;
}
int main (int argc, char **argv) {
char *url;
int brate = 32;
int c;
while ((c = getopt(argc, argv, "u:b:")) != -1)
switch (c) {
case 'u':
url = strdup(optarg);
break;
case 'b':
brate = atoi(optarg);
break;
default:
abort();
}
// libev
loop = ev_default_loop(0);
ev_signal_init(&sigint_sig, sigint_cb, SIGINT);
ev_signal_start (loop, &sigint_sig);
ev_async_init(&async, send_cb);
ev_async_start(loop, &async);
// jack
jack_status_t status;
ringbuf = jack_ringbuffer_create(buffer_size);
framebuf = malloc(buffer_samples * sizeof(jack_default_audio_sample_t));
client = jack_client_open("jack2rtmp", JackNullOption, &status);
if (client == NULL) {
fprintf(stderr, "jack server not running?\n");
return EXIT_FAILURE;
}
jack_on_shutdown(client, jack_shutdown, NULL);
jack_set_process_callback(client, jack_callback, NULL);
//jack_set_latency_callback(client, jack_latency_callback, NULL);
output_port = jack_port_register(client, "1", JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
//jack_port_register(client, "2", JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0);
if (jack_activate(client)) {
fprintf(stderr, "cannot activate client");
}
// lame
mp3 = lame_init();
lame_set_num_channels(mp3, 1);
lame_set_mode(mp3, MONO);
lame_set_in_samplerate(mp3, 48000);
lame_set_out_samplerate(mp3, 44100);
lame_set_quality(mp3, 2); // 0=best (very slow), 9=worst.
lame_set_brate(mp3, brate);
lame_init_params(mp3);
// rtmp
rtmp_o = RTMP_Alloc();
if (rtmp_o == NULL) {
fprintf(stderr, "RTMP_Alloc\n");
return EXIT_FAILURE;
}
RTMP_Init(rtmp_o);
RTMP_SetupURL(rtmp_o, url);
RTMP_EnableWrite(rtmp_o);
if (!RTMP_Connect(rtmp_o, NULL) || !RTMP_ConnectStream(rtmp_o, 0)) {
RTMP_Free(rtmp_o);
fprintf(stderr, "Can not connect.\n");
return EXIT_FAILURE;
} else {
fprintf(stderr, "Connected.\n");
}
/*
char *rtmp_buffer = malloc(4096);
char *rtmp_top = rtmp_buffer + RTMP_MAX_HEADER_SIZE;
char *outend = rtmp_buffer + 4096;
*rtmp_top = (uint8_t) AMF_STRING;
rtmp_top++;
rtmp_top = put_amf_string(rtmp_top, "@setDataFrame");
*rtmp_top = (uint8_t) AMF_STRING;
rtmp_top++;
rtmp_top = put_amf_string(rtmp_top, "onMetaData");
// описание объекта
*rtmp_top = (uint8_t) 8;
rtmp_top++;
rtmp_top = AMF_EncodeInt32(rtmp_top, outend, 8);
// содержимое объекта
rtmp_top = put_amf_string(rtmp_top, "duration");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 0);
rtmp_top = put_amf_string(rtmp_top, "audiodatarate");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, brate);
rtmp_top = put_amf_string(rtmp_top, "audiosamplerate");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 44100);
rtmp_top = put_amf_string(rtmp_top, "audiosamplesize");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 16);
rtmp_top = put_amf_string(rtmp_top, "stereo");
rtmp_top = AMF_EncodeBoolean(rtmp_top, outend, FALSE);
rtmp_top = put_amf_string(rtmp_top, "audiocodecid");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 2);
rtmp_top = put_amf_string(rtmp_top, "encoder");
*rtmp_top = (uint8_t) AMF_STRING;
rtmp_top++;
rtmp_top = put_amf_string(rtmp_top, "Lavf53.21.1");
rtmp_top = put_amf_string(rtmp_top, "filesize");
rtmp_top = AMF_EncodeNumber(rtmp_top, outend, 0);
// конец объекта
rtmp_top = put_amf_string(rtmp_top, "");
*rtmp_top = (uint8_t) AMF_OBJECT_END;
rtmp_top++;
rtmp_w.m_headerType = RTMP_PACKET_SIZE_LARGE;
rtmp_w.m_nChannel = 0x04; // source channel
rtmp_w.m_nBodySize = rtmp_top - rtmp_buffer - RTMP_MAX_HEADER_SIZE;
rtmp_w.m_packetType = RTMP_PACKET_TYPE_INFO;
rtmp_w.m_nInfoField2 = rtmp_o->m_stream_id;
rtmp_w.m_body = rtmp_buffer + RTMP_MAX_HEADER_SIZE;
if ( !RTMP_SendPacket(rtmp_o, &rtmp_w, TRUE) ) {
fprintf(stderr, "RTMP_SendPacket error\n");
}
free(rtmp_buffer);
*/
int off = 0;
setsockopt(RTMP_Socket(rtmp_o), SOL_TCP, TCP_NODELAY, &off, sizeof(off));
rtmp_i = TRUE;
ev_loop(loop, 0);
jack_ringbuffer_free(ringbuf);
//jack_client_close(client);
return EXIT_SUCCESS;
}
void doServe(STREAMING_SERVER * server, // server socket and state (our listening socket)
int sockfd // client connection socket
)
{
server->state = STREAMING_IN_PROGRESS;
RTMP *rtmp = RTMP_Alloc(); /* our session with the real client */
RTMPPacket packet = { 0 };
// timeout for http requests
fd_set fds;
struct timeval tv;
memset(&tv, 0, sizeof(struct timeval));
tv.tv_sec = 5;
FD_ZERO(&fds);
FD_SET(sockfd, &fds);
if (select(sockfd + 1, &fds, NULL, NULL, &tv) <= 0)
{
RTMP_Log(RTMP_LOGERROR, "Request timeout/select failed, ignoring request");
goto quit;
}
else
{
RTMP_Init(rtmp);
rtmp->m_sb.sb_socket = sockfd;
if (sslCtx && !RTMP_TLS_Accept(rtmp, sslCtx))
{
RTMP_Log(RTMP_LOGERROR, "TLS handshake failed");
goto cleanup;
}
if (!RTMP_Serve(rtmp))
{
RTMP_Log(RTMP_LOGERROR, "Handshake failed");
goto cleanup;
}
}
server->arglen = 0;
while (RTMP_IsConnected(rtmp) && RTMP_ReadPacket(rtmp, &packet))
{
if (!RTMPPacket_IsReady(&packet))
continue;
ServePacket(server, rtmp, &packet);
RTMPPacket_Free(&packet);
}
cleanup:
RTMP_LogPrintf("Closing connection... ");
RTMP_Close(rtmp);
/* Should probably be done by RTMP_Close() ... */
rtmp->Link.playpath.av_val = NULL;
rtmp->Link.tcUrl.av_val = NULL;
rtmp->Link.swfUrl.av_val = NULL;
rtmp->Link.pageUrl.av_val = NULL;
rtmp->Link.app.av_val = NULL;
rtmp->Link.flashVer.av_val = NULL;
if (rtmp->Link.usherToken.av_val)
{
free(rtmp->Link.usherToken.av_val);
rtmp->Link.usherToken.av_val = NULL;
}
RTMP_Free(rtmp);
RTMP_LogPrintf("done!\n\n");
quit:
if (server->state == STREAMING_IN_PROGRESS)
server->state = STREAMING_ACCEPTING;
return;
}
