Adding MP3 encoding using lame as an option

inc/ALSACapture.h

@@ -18,21 +18,29 @@
 
 #include <alsa/asoundlib.h>
 #include "logger.h"
+#include "AudioCompressor.h"
+
+
+#define COMPRESSED_AUDIO_FMT_NONE 0
+#define COMPRESSED_AUDIO_FMT_MP3 1
 
 #include "DeviceInterface.h"
 
 struct ALSACaptureParameters 
 {
-	ALSACaptureParameters(const char* devname, const std::list<snd_pcm_format_t> & formatList, unsigned int sampleRate, unsigned int channels, int verbose) : 
-		m_devName(devname), m_formatList(formatList), m_sampleRate(sampleRate), m_channels(channels), m_verbose(verbose) {
-
+	ALSACaptureParameters(const char* devname, const std::list<snd_pcm_format_t> & formatList,
+		unsigned int sampleRate, unsigned int channels, int verbose, AudioCompressor* audioCompressor) : 
+		m_devName(devname), m_formatList(formatList), m_sampleRate(sampleRate), m_channels(channels),
+		m_verbose(verbose), m_audioCompressor(audioCompressor) {
+
 	}
 
 	std::string      m_devName;
 	std::list<snd_pcm_format_t> m_formatList;		
 	unsigned int     m_sampleRate;
 	unsigned int     m_channels;
 	int              m_verbose;
+	AudioCompressor* m_audioCompressor;
 };
 
 class ALSACapture  : public DeviceInterface
@@ -61,6 +69,8 @@ class ALSACapture  : public DeviceInterface
 		unsigned long         m_periodSize;
 		ALSACaptureParameters m_params;
 		snd_pcm_format_t      m_fmt;
+
+
 };
 
 #endif

inc/AudioCompressor.h

@@ -0,0 +1,22 @@
+/* ---------------------------------------------------------------------------
+** This software is in the public domain, furnished "as is", without technical
+** support, and with no warranty, express or implied, as to its usefulness for
+** any purpose.
+**
+** AudioCompressor.h
+** 
+** Contains abstract class for audio compressors
+**
+** -------------------------------------------------------------------------*/
+
+#pragma once
+
+#include "logger.h"
+
+class AudioCompressor
+{
+	public:
+		AudioCompressor();
+		virtual int compress(short* pcm_data, int sample_count, char* output_buffer, int output_buffer_size) { return -1; };
+	private:
+};

inc/MP3AudioCompressor.h

@@ -0,0 +1,26 @@
+/* ---------------------------------------------------------------------------
+** This software is in the public domain, furnished "as is", without technical
+** support, and with no warranty, express or implied, as to its usefulness for
+** any purpose.
+**
+** MP3AudioCompressor.h
+** 
+** Contains abstract class for audio compressors
+**
+** -------------------------------------------------------------------------*/
+
+#pragma once
+
+#include "logger.h"
+#include <lame.h>
+#include "AudioCompressor.h"
+
+class MP3AudioCompressor: public AudioCompressor
+{
+	public:
+		MP3AudioCompressor();
+		MP3AudioCompressor(int channels, int samplerate);
+		int compress(short* pcm_data, int sample_count, char* output_buffer, int output_buffer_size);		
+	private:
+		lame_global_flags* gfp;
+};

inc/NullAudioCompressor.h

@@ -0,0 +1,18 @@
+/* ---------------------------------------------------------------------------
+** This software is in the public domain, furnished "as is", without technical
+** support, and with no warranty, express or implied, as to its usefulness for
+** any purpose.
+** -------------------------------------------------------------------------*/
+
+#pragma once
+
+#include "logger.h"
+#include "AudioCompressor.h"
+
+class NullAudioCompressor: public AudioCompressor
+{
+	public:
+		NullAudioCompressor();
+		int compress(short* pcm_data, int sample_count, char* output_buffer, int output_buffer_size);		
+	private:
+};

inc/ServerMediaSubsession.h

@@ -36,16 +36,20 @@ class BaseServerMediaSubsession
 {
 	public:
 		BaseServerMediaSubsession(StreamReplicator* replicator): m_replicator(replicator) {
+
 			V4L2DeviceSource* deviceSource = dynamic_cast<V4L2DeviceSource*>(replicator->inputSource());
+
 			if (deviceSource) {
                 DeviceInterface* device = deviceSource->getDevice();
                 if (device->getVideoFormat() >= 0) {
     	            m_format = BaseServerMediaSubsession::getVideoRtpFormat(device->getVideoFormat());		
                 } else {
-    	            m_format = BaseServerMediaSubsession::getAudioRtpFormat(device->getAudioFormat(), device->getSampleRate(), device->getChannels());						
+    	            m_format = BaseServerMediaSubsession::getAudioRtpFormat(device->getAudioFormat(), device->getSampleRate(), device->getChannels());
                 }
                 LOG(NOTICE) << "format:" << m_format;
 			}
+
+
 	    }
 
 		// -----------------------------------------
@@ -69,9 +73,13 @@ class BaseServerMediaSubsession
             return rtpFormat;
         }
 
+        /*
+         * Return the format for the audio part of the RTP session.
+         */
         static std::string getAudioRtpFormat(int format, int sampleRate, int channels)
         {
             std::ostringstream os;
+
 #ifdef HAVE_ALSA            
             os << "audio/";
             switch (format) {                
@@ -100,6 +108,8 @@ class BaseServerMediaSubsession
                     break;
             }
             os << "/" << sampleRate << "/" << channels;
+
+
 #endif            
             return os.str();
         }        
@@ -111,6 +121,8 @@ class BaseServerMediaSubsession
 
 	protected:
 		StreamReplicator* m_replicator;
-		std::string m_format;        
+		std::string m_format;
+        int m_compressedAudioFmt;
+
 };
 

inc/UnicastServerMediaSubsession.h

@@ -18,11 +18,12 @@ class UnicastServerMediaSubsession : public OnDemandServerMediaSubsession , publ
 {
 	public:
 		static UnicastServerMediaSubsession* createNew(UsageEnvironment& env, StreamReplicator* replicator);
+		static UnicastServerMediaSubsession* createNew(UsageEnvironment& env, StreamReplicator* replicator, int compressedAudioFmt);
 
 	protected:
 		UnicastServerMediaSubsession(UsageEnvironment& env, StreamReplicator* replicator) 
 				: OnDemandServerMediaSubsession(env, False), BaseServerMediaSubsession(replicator) {}
-			
+
 		virtual FramedSource* createNewStreamSource(unsigned clientSessionId, unsigned& estBitrate);
 		virtual RTPSink* createNewRTPSink(Groupsock* rtpGroupsock,  unsigned char rtpPayloadTypeIfDynamic, FramedSource* inputSource);		
 		virtual char const* getAuxSDPLine(RTPSink* rtpSink,FramedSource* inputSource);	

inc/V4l2RTSPServer.h

@@ -22,6 +22,7 @@
 #include "MulticastServerMediaSubsession.h"
 #include "TSServerMediaSubsession.h"
 #include "HTTPServer.h"
+#include "AudioCompressor.h"
 
 class V4l2RTSPServer {
     public:
@@ -124,13 +125,13 @@ class V4l2RTSPServer {
 #ifdef HAVE_ALSA
         StreamReplicator* CreateAudioReplicator(
 			const std::string& audioDev, const std::list<snd_pcm_format_t>& audioFmtList, int audioFreq, int audioNbChannels, int verbose,
-			int queueSize, int useThread);
+			int queueSize, int useThread, AudioCompressor* compressor);
 #endif
 
         // -----------------------------------------
         //    Add unicast Session
         // -----------------------------------------
-        int AddUnicastSession(const std::string& url, StreamReplicator* videoReplicator, StreamReplicator* audioReplicator) {
+        int AddUnicastSession(const std::string& url, StreamReplicator* videoReplicator, StreamReplicator* audioReplicator, int compressedAudioFmt) {
 			// Create Unicast Session					
 			std::list<ServerMediaSubsession*> subSession;
 			if (videoReplicator)
@@ -139,7 +140,7 @@ class V4l2RTSPServer {
 			}
 			if (audioReplicator)
 			{
-				subSession.push_back(UnicastServerMediaSubsession::createNew(*this->env(), audioReplicator));				
+				subSession.push_back(UnicastServerMediaSubsession::createNew(*this->env(), audioReplicator, compressedAudioFmt));				
 			}
 			return this->addSession(url, subSession);	    
         }

main.cpp

@@ -33,6 +33,9 @@
 
 #include "DeviceSourceFactory.h"
 #include "V4l2RTSPServer.h"
+#include "AudioCompressor.h"
+#include "NullAudioCompressor.h"
+#include "MP3AudioCompressor.h"
 
 
 // -----------------------------------------
@@ -156,6 +159,12 @@ int main(int argc, char** argv)
 	int audioNbChannels = 2;
 	std::list<snd_pcm_format_t> audioFmtList;
 	snd_pcm_format_t audioFmt = SND_PCM_FORMAT_UNKNOWN;
+	int compressedAudioFmt = COMPRESSED_AUDIO_FMT_NONE;
+
+	AudioCompressor *compressor;
+	std::list<AudioCompressor *> audioCompressors;
+
+
 #endif	
 	const char* defaultPort = getenv("PORT");
 	if (defaultPort != NULL) {
@@ -164,7 +173,7 @@ int main(int argc, char** argv)
 
 	// decode parameters
 	int c = 0;     
-	while ((c = getopt (argc, argv, "v::Q:O:b:" "I:P:p:m::u:M:ct:S::" "R:U:" "rwBsf::F:W:H:G:" "A:C:a:" "Vh")) != -1)
+	while ((c = getopt (argc, argv, "v::Q:O:b:" "I:P:p:m::u:M:ct:S::" "R:U:" "rwBsf::F:W:H:G:" "A:C:a:l:" "Vh")) != -1)
 	{
 		switch (c)
 		{
@@ -204,6 +213,10 @@ int main(int argc, char** argv)
 			case 'A':	audioFreq = atoi(optarg); break;
 			case 'C':	audioNbChannels = atoi(optarg); break;
 			case 'a':	audioFmt = decodeAudioFormat(optarg); if (audioFmt != SND_PCM_FORMAT_UNKNOWN) {audioFmtList.push_back(audioFmt);} ; break;
+
+			case 'l':	compressedAudioFmt = atoi(optarg); break;
+
+
 #endif			
 
 			// version
@@ -255,6 +268,8 @@ int main(int argc, char** argv)
 				std::cout << "\t -A freq          : ALSA capture frequency and channel (default " << audioFreq << ")"                                << std::endl;
 				std::cout << "\t -C channels      : ALSA capture channels (default " << audioNbChannels << ")"                                       << std::endl;
 				std::cout << "\t -a fmt           : ALSA capture audio format (default S16_BE)"                                                      << std::endl;
+				std::cout << "\t -l [number]      : Compresssed audio format: 0 - None, 1 - MP3"                                                                 << std::endl;
+
 #endif
 
 				std::cout << "\t Devices :"                                                                                                    << std::endl;
@@ -342,9 +357,26 @@ int main(int argc, char** argv)
 			// Init Audio Capture
 			StreamReplicator* audioReplicator = NULL;
 #ifdef HAVE_ALSA
+
+
+
+			if (compressedAudioFmt == COMPRESSED_AUDIO_FMT_NONE) {
+				compressor = new NullAudioCompressor();
+				audioCompressors.push_back(compressor);
+			}
+
+			if (compressedAudioFmt == COMPRESSED_AUDIO_FMT_MP3) {
+				compressor = new MP3AudioCompressor(audioNbChannels, audioFreq);
+				audioCompressors.push_back(compressor);
+			}
+
+			LOG(NOTICE) << "Using AudioCompressor: " << compressor;		
+
+
 			audioReplicator = rtspServer.CreateAudioReplicator(
 					audioDev, audioFmtList, audioFreq, audioNbChannels, verbose,
-					queueSize, useThread);		
+					queueSize, useThread, compressor);
+
 #endif
 
 
@@ -361,7 +393,8 @@ int main(int argc, char** argv)
 			}
 
 			// Create Unicast Session
-			nbSource += rtspServer.AddUnicastSession(baseUrl+url, videoReplicator, audioReplicator);		
+			nbSource += rtspServer.AddUnicastSession(baseUrl+url, videoReplicator, audioReplicator, compressedAudioFmt);		
+
 		}
 
 		if (nbSource>0)