Add audio to all stream (thanks to @PieVo)

src/rRTSPServer/Makefile.rRTSPServer

@@ -72,7 +72,8 @@ livemedia:
 rRTSPServer_OBJS	= src/rRTSPServer.$(OBJ) \
 				src/ByteStreamCBMemorySource.$(OBJ) src/H264VideoCBMemoryServerMediaSubsession.$(OBJ) \
 				src/CBMemoryServerMediaSubsession.$(OBJ) \
-				src/WAVAudioFifoSource.$(OBJ) src/WAVAudioFifoServerMediaSubsession.$(OBJ)
+				src/WAVAudioFifoSource.$(OBJ) src/WAVAudioFifoServerMediaSubsession.$(OBJ) \
+				src/DummySink.$(OBJ)
 
 rRTSPServer$(EXE):	$(rRTSPServer_OBJS) $(LOCAL_LIBS)
 	$(LINK)$@ $(CONSOLE_LINK_OPTS) $(rRTSPServer_OBJS) $(LIBS) -lpthread

src/rRTSPServer/include/DummySink.hh

@@ -0,0 +1,29 @@
+#include "liveMedia.hh"
+#include "BasicUsageEnvironment.hh"
+
+class DummySink: public MediaSink {
+public:
+  static DummySink* createNew(UsageEnvironment& env,
+			      char const* streamId = NULL); // identifies the stream itself (optional)
+
+private:
+  DummySink(UsageEnvironment& env, char const* streamId);
+    // called only by "createNew()"
+  virtual ~DummySink();
+
+  static void afterGettingFrame(void* clientData, unsigned frameSize,
+                                unsigned numTruncatedBytes,
+				struct timeval presentationTime,
+                                unsigned durationInMicroseconds);
+  void afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
+			 struct timeval presentationTime, unsigned durationInMicroseconds);
+
+private:
+  // redefined virtual functions:
+  virtual Boolean continuePlaying();
+
+private:
+  u_int8_t* fReceiveBuffer;
+  char* fStreamId;
+};
+

src/rRTSPServer/include/WAVAudioFifoServerMediaSubsession.hh

@@ -25,17 +25,15 @@ along with this library; if not, write to the Free Software Foundation, Inc.,
 #ifndef _FILE_SERVER_MEDIA_SUBSESSION_HH
 #include "FileServerMediaSubsession.hh"
 #endif
+#include "StreamReplicator.hh"
 
-class WAVAudioFifoServerMediaSubsession: public FileServerMediaSubsession{
+class WAVAudioFifoServerMediaSubsession: public OnDemandServerMediaSubsession {
 public:
   static WAVAudioFifoServerMediaSubsession*
-  createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource,
-	    Boolean convertToULaw = False);
-      // If "convertToULaw" is True, 16-bit audio streams are converted to
-      // 8-bit u-law audio prior to streaming.
+  createNew(UsageEnvironment& env, StreamReplicator* replicator, Boolean reuseFirstSource, Boolean convertToULaw);
 
 protected:
-  WAVAudioFifoServerMediaSubsession(UsageEnvironment& env, char const* fileName,
+  WAVAudioFifoServerMediaSubsession(UsageEnvironment& env, StreamReplicator* replicator,
 				    Boolean reuseFirstSource, Boolean convertToULaw);
       // called only by createNew();
   virtual ~WAVAudioFifoServerMediaSubsession();
@@ -54,7 +52,6 @@ protected: // redefined virtual functions
   virtual float duration() const;
 
 protected:
-  Boolean fConvertToULaw;
 
   // The following parameters of the input stream are set after
   // "createNewStreamSource" is called:
@@ -63,6 +60,8 @@ protected:
   unsigned fSamplingFrequency;
   unsigned fNumChannels;
   float fFileDuration;
+  StreamReplicator* fReplicator;
+  Boolean fConvertToULaw;
 };
 
 #endif

src/rRTSPServer/include/WAVAudioFifoSource.hh

@@ -27,14 +27,11 @@ along with this library; if not, write to the Free Software Foundation, Inc.,
 #include "AudioInputDevice.hh"
 #endif
 
-typedef enum {
-  WA_PCM = 0x01,
-  WA_PCMA = 0x06,
-  WA_PCMU = 0x07,
-  WA_IMA_ADPCM = 0x11,
-  WA_UNKNOWN
-} WAV_AUDIO_FORMAT;
-
+#define  WA_PCM 0x01
+#define  WA_PCMA 0x06
+#define  WA_PCMU 0x07
+#define  WA_IMA_ADPCM 0x11
+#define  WA_UNKNOWN 0x12
 
 class WAVAudioFifoSource: public AudioInputDevice {
 public:

src/rRTSPServer/include/rRTSPServer.h

@@ -48,8 +48,8 @@
 #define RESOLUTION_HIGH 1080
 #define RESOLUTION_BOTH 1440
 
-#define OUTPUT_BUFFER_SIZE_LOW  32768
-#define OUTPUT_BUFFER_SIZE_HIGH 131072
+#define OUTPUT_BUFFER_SIZE_LOW  49152
+#define OUTPUT_BUFFER_SIZE_HIGH 196608
 
 typedef struct
 {

src/rRTSPServer/src/DummySink.cpp

@@ -0,0 +1,58 @@
+#include "liveMedia.hh"
+#include "BasicUsageEnvironment.hh"
+#include "DummySink.hh"
+
+// Implementation of "DummySink":
+
+// Even though we're not going to be doing anything with the incoming data, we still need to receive it.
+// Define the size of the buffer that we'll use:
+#define DUMMY_SINK_RECEIVE_BUFFER_SIZE 100000
+
+DummySink* DummySink::createNew(UsageEnvironment& env, char const* streamId) {
+  return new DummySink(env, streamId);
+}
+
+DummySink::DummySink(UsageEnvironment& env, char const* streamId)
+  : MediaSink(env) {
+  fStreamId = strDup(streamId);
+  fReceiveBuffer = new u_int8_t[DUMMY_SINK_RECEIVE_BUFFER_SIZE];
+}
+
+DummySink::~DummySink() {
+  delete[] fReceiveBuffer;
+  delete[] fStreamId;
+}
+
+void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,
+				  struct timeval presentationTime, unsigned durationInMicroseconds) {
+  DummySink* sink = (DummySink*)clientData;
+  sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);
+}
+
+// If you don't want to see debugging output for each received frame, then comment out the following line:
+#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1
+
+void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,
+				  struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {
+  // We've just received a frame of data.  (Optionally) print out information about it:
+#ifdef DEBUG
+  if ((int)presentationTime.tv_sec % 10 == 0) {
+    char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time
+    sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
+    envir() << "DummySink Presentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;
+  }
+#endif 
+
+  // Then continue, to request the next frame of data:
+  continuePlaying();
+}
+
+Boolean DummySink::continuePlaying() {
+  if (fSource == NULL) return False; // sanity check (should not happen)
+
+  // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:
+  fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,
+                        afterGettingFrame, this,
+                        onSourceClosure, this);
+  return True;
+}

src/rRTSPServer/src/WAVAudioFifoServerMediaSubsession.cpp

@@ -25,16 +25,14 @@ along with this library; if not, write to the Free Software Foundation, Inc.,
 #include "SimpleRTPSink.hh"
 
 WAVAudioFifoServerMediaSubsession* WAVAudioFifoServerMediaSubsession
-::createNew(UsageEnvironment& env, char const* fileName, Boolean reuseFirstSource,
-	    Boolean convertToULaw) {
-  return new WAVAudioFifoServerMediaSubsession(env, fileName,
-					       reuseFirstSource, convertToULaw);
+::createNew(UsageEnvironment& env, StreamReplicator* replicator, Boolean reuseFirstSource, Boolean convertToULaw) {
+  return new WAVAudioFifoServerMediaSubsession(env, replicator, reuseFirstSource, convertToULaw);
 }
 
 WAVAudioFifoServerMediaSubsession
-::WAVAudioFifoServerMediaSubsession(UsageEnvironment& env, char const* fileName,
-				    Boolean reuseFirstSource, Boolean convertToULaw)
-  : FileServerMediaSubsession(env, fileName, reuseFirstSource),
+::WAVAudioFifoServerMediaSubsession(UsageEnvironment& env, StreamReplicator* replicator, Boolean reuseFirstSource, Boolean convertToULaw)
+  : OnDemandServerMediaSubsession(env, reuseFirstSource),
+    fReplicator(replicator),
     fConvertToULaw(convertToULaw) {
 }
 
@@ -97,51 +95,49 @@ ::setStreamSourceScale(FramedSource* inputSource, float scale) {
 FramedSource* WAVAudioFifoServerMediaSubsession
 ::createNewStreamSource(unsigned /*clientSessionId*/, unsigned& estBitrate) {
   FramedSource* resultSource = NULL;
-  do {
-    WAVAudioFifoSource* wavSource = WAVAudioFifoSource::createNew(envir(), fFileName);
-    if (wavSource == NULL) break;
-
-    // Get attributes of the audio source:
-
-    fAudioFormat = wavSource->getAudioFormat();
-    fBitsPerSample = wavSource->bitsPerSample();
-    // We handle only 4,8,16,20,24 bits-per-sample audio:
-    if (fBitsPerSample%4 != 0 || fBitsPerSample < 4 || fBitsPerSample > 24 || fBitsPerSample == 12) {
-      envir() << "The input file contains " << fBitsPerSample << " bit-per-sample audio, which we don't handle\n";
-      break;
+  WAVAudioFifoSource* originalSource = NULL;
+  FramedFilter* previousSource = (FramedFilter*)fReplicator->inputSource();
+
+  // Iterate back into the filter chain until a source is found that 
+  // has a sample frequency and expected to be a WAVAudioFifoSource.
+  for (int x = 0; x < 10; x++) {
+    if (((WAVAudioFifoSource*)(previousSource))->bitsPerSample() != 0) {
+#ifdef DEBUG
+      printf("WAVAudioFifoSource found at x = %d\n", x);
+#endif
+      originalSource = (WAVAudioFifoSource*)(previousSource);
+      break; 
     }
-    fSamplingFrequency = wavSource->samplingFrequency();
-    fNumChannels = wavSource->numChannels();
+    previousSource = (FramedFilter*)previousSource->inputSource();
+  }
+#ifdef DEBUG
+  printf("fReplicator->inputSource() = %p\n", originalSource);
+#endif
+  resultSource = fReplicator->createStreamReplica();
+  if (resultSource == NULL) {
+    printf("Failed to create stream replica\n");
+    Medium::close(resultSource);
+    return NULL;
+  }
+  else {
+    fAudioFormat = originalSource->getAudioFormat();
+    fBitsPerSample = originalSource->bitsPerSample();
+    fSamplingFrequency = originalSource->samplingFrequency();
+    fConvertToULaw = True;
+    fNumChannels = originalSource->numChannels();
     unsigned bitsPerSecond = fSamplingFrequency*fBitsPerSample*fNumChannels;
+#ifdef DEBUG    
+    printf("Original source FMT: %d bps: %d freq: %d\n", fAudioFormat, fBitsPerSample, fSamplingFrequency);
+#endif
+    fFileDuration = ~0;//(float)((8.0*originalSource->numPCMBytes())/(fSamplingFrequency*fNumChannels*fBitsPerSample));
 
-    fFileDuration = (float)((8.0*wavSource->numPCMBytes())/(fSamplingFrequency*fNumChannels*fBitsPerSample));
+    estBitrate = (bitsPerSecond+500)/1000; // kbps
 
-    // Add in any filter necessary to transform the data prior to streaming:
-    resultSource = wavSource; // by default
-    if (fAudioFormat == WA_PCM) {
-      if (fBitsPerSample == 16) {
-	// Note that samples in the WAV audio file are in little-endian order.
-	if (fConvertToULaw) {
-	  // Add a filter that converts from raw 16-bit PCM audio to 8-bit u-law audio:
-	  resultSource = uLawFromPCMAudioSource::createNew(envir(), wavSource, 1/*little-endian*/);
-	  bitsPerSecond /= 2;
-	} else {
-	  // Add a filter that converts from little-endian to network (big-endian) order:
-	  resultSource = EndianSwap16::createNew(envir(), wavSource);
-	}
-      } else if (fBitsPerSample == 20 || fBitsPerSample == 24) {
-	// Add a filter that converts from little-endian to network (big-endian) order:
-	resultSource = EndianSwap24::createNew(envir(), wavSource);
-      }
-    }
+    if (fConvertToULaw)
+        estBitrate /= 2;
 
-    estBitrate = (bitsPerSecond+500)/1000; // kbps
     return resultSource;
-  } while (0);
-
-  // An error occurred:
-  Medium::close(resultSource);
-  return NULL;
+  }
 }
 
 RTPSink* WAVAudioFifoServerMediaSubsession
@@ -200,7 +196,9 @@ ::createNewRTPSink(Groupsock* rtpGroupsock,
     } else { //unknown format
       break;
     }
-
+#ifdef DEBUG
+    printf("Create SimpleRTPSink: %s, freq: %d, channels %d\n", mimeType, fSamplingFrequency, fNumChannels);
+#endif
     return SimpleRTPSink::createNew(envir(), rtpGroupsock,
 				    payloadFormatCode, fSamplingFrequency,
 				    "audio", mimeType, fNumChannels);