work on bypass-audio.

CALLSTATUS monitor message has gained a field, thus changing
API for java side.

Android.mk

@@ -71,7 +71,8 @@ LOCAL_SRC_FILES:= \
 	serval-dna/codecs.c \
 	serval-dna/audiodevices.c \
 	serval-dna/audio_msm_g1.c \
-	serval-dna/audio_alsa.c
+	serval-dna/audio_alsa.c \
+	serval-dna/audio_reflector.c
 
 LOCAL_MODULE:= serval
 

Makefile.in

@@ -44,7 +44,8 @@ SRCS=	batman.c \
 	codecs.c \
 	audiodevices.c \
 	audio_msm_g1.c \
-	audio_alsa.c
+	audio_alsa.c \
+	audio_reflector.c
 
 HAVE_VOIPTEST=	@HAVE_VOIPTEST@
 ifeq ($(HAVE_VOIPTEST), 1)

audio_alsa.c

@@ -18,7 +18,13 @@
 
 #include "serval.h"
 
-#ifdef HAVE_ALSA_ASOUNDLIB_H
+#ifndef HAVE_ALSA_ASOUNDLIB_H
+/* XXX Android NDK lacks the header files needed to build this.
+   Will deal with it later */
+monitor_audio *audio_alsa_detect() {
+  return NULL;
+}
+#else
 
 #include <stdio.h>
 #include <stdlib.h>
@@ -40,7 +46,7 @@
    necessary symbols we need.
 */
 typedef struct alsa_functions {
-  int (*snd_pcm_open)(snd_pcm_t **,char *,int,int);
+  int (*snd_pcm_open_preferred)(snd_pcm_t **,int *,int *,int);
   int (*snd_pcm_close)(snd_pcm_t *);
   int (*snd_pcm_hw_params_malloc)(snd_pcm_hw_params_t **);
   int (*snd_pcm_hw_params_any)(snd_pcm_t *,snd_pcm_hw_params_t *);
@@ -63,10 +69,12 @@ alsa_functions *alsa = NULL;
 
 int alsa_load()
 {
+  WHY("Trying to load ALSA library");
   void *h = dlopen(ALSA_LIB_PATH,RTLD_LAZY);
+  if (!h) h=dlopen("/usr/lib/i386-linux-gnu/libasound.so.2",RTLD_LAZY);
   if (!h) return -1;
   alsa_functions *a=calloc(sizeof(alsa_functions),1);
-  GETSYM(snd_pcm_open);
+  GETSYM(snd_pcm_open_preferred);
   GETSYM(snd_pcm_hw_params_malloc);
   GETSYM(snd_pcm_hw_params_any);
   GETSYM(snd_pcm_hw_params_set_access);
@@ -81,7 +89,8 @@ int alsa_load()
   GETSYM(snd_pcm_poll_descriptors);
   alsa=a;
   dlclose(h);
-  
+  WHY("Loaded ALSA library");
+
   return 0;
 }
 
@@ -120,8 +129,8 @@ int audio_alsa_start_play()
   record_params=NULL; play_params=NULL;
 
   /* Open playback device */
-  r = alsa->snd_pcm_open (&play_handle,"default",SND_PCM_STREAM_PLAYBACK,
+  r = alsa->snd_pcm_open_preferred (&play_handle,NULL,NULL,
-			  SND_PCM_NONBLOCK);
+				    SND_PCM_STREAM_PLAYBACK|SND_PCM_NONBLOCK);
   if (r) { WHYF("ALSA pcm_open() failed"); goto error; }
 
   /* Configure playback device for 8000Hz, 16 bit, mono */
@@ -157,8 +166,8 @@ int audio_alsa_start_play()
 int audio_alsa_start_record()
 {
   /* Open recording device non-blocking */
-  int r = alsa->snd_pcm_open (&record_handle,"default",SND_PCM_STREAM_CAPTURE,
+  int r = alsa->snd_pcm_open_preferred(&record_handle,NULL,NULL,
-			      SND_PCM_NONBLOCK);
+				       SND_PCM_STREAM_CAPTURE|SND_PCM_NONBLOCK);
   if (r) { WHYF("ALSA pcm_open() failed"); goto error; }
 
   /* Configure playback device for 8000Hz, 16 bit, mono */
@@ -243,25 +252,23 @@ int audio_alsa_write(unsigned char *buffer,int bytes)
   else return 0;
 }
 
-#endif // HAVE_ALSA_ASOUNDLIB_H
-
 monitor_audio *audio_alsa_detect()
 {
-#ifdef HAVE_ALSA_ASOUNDLIB_H
   if (!alsa) alsa_load();
   if (!alsa) return NULL;
   snd_pcm_t *handle;
-  if (alsa->snd_pcm_open (&handle, "default", SND_PCM_STREAM_PLAYBACK, 0) != -1) {
+  if (alsa->snd_pcm_open_preferred(&handle,NULL,NULL,SND_PCM_STREAM_PLAYBACK) < 0)
-    alsa->snd_pcm_close(handle);
+    return NULL;
-    monitor_audio *au=calloc(sizeof(monitor_audio),1);
+  alsa->snd_pcm_close(handle);
-    strcpy(au->name,"ALSA compatible");
+
-    au->start=audio_alsa_start;
+  monitor_audio *au=calloc(sizeof(monitor_audio),1);
-    au->stop=audio_alsa_stop;
+  strcpy(au->name,"ALSA compatible");
-    au->poll_fds=audio_alsa_pollfds;
+  au->start=audio_alsa_start;
-    au->read=audio_alsa_read;
+  au->stop=audio_alsa_stop;
-    au->write=audio_alsa_write;
+  au->poll_fds=audio_alsa_pollfds;
-    return au;
+  au->read=audio_alsa_read;
-  }
+  au->write=audio_alsa_write;
-#endif // HAVE_ALSA_ASOUNDLIB_H
+  return au;
-  return NULL;
 }
+
+#endif

audio_reflector.c

@@ -0,0 +1,57 @@
+/* 
+   Copyright (C) 2012 Paul Gardner-Stephen 
+   
+   This program is free software; you can redistribute it and/or
+   modify it under the terms of the GNU General Public License
+   as published by the Free Software Foundation; either version 2
+   of the License, or (at your option) any later version.
+   
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+   
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301, USA.
+*/
+
+#include "serval.h"
+
+int audio_reflector_null() { return 0; }
+int audio_reflector_pollfds(struct pollfd *fd,int slots) { return 0; }
+
+unsigned char reflectBuffer[8192];
+int reflectBufferLen=0;
+
+int audio_reflector_write(unsigned char *buffer,int len)
+{
+  if (reflectBufferLen+len>=8192) 
+    len=8192-reflectBufferLen;
+  bcopy(&buffer[0],&reflectBuffer[reflectBufferLen],len);
+  reflectBufferLen+=len;
+  return len;
+}
+
+int audio_reflector_read(unsigned char *buffer,int maximum_count)
+{
+  int count=reflectBufferLen;
+  if (count>maximum_count) count=maximum_count;
+  bcopy(&reflectBuffer[0],&buffer[0],count);
+  bcopy(&reflectBuffer[count],&reflectBuffer[0],
+	reflectBufferLen-count);
+  reflectBufferLen-=count;
+  return count;
+}
+
+monitor_audio *audio_reflector_detect()
+{
+  monitor_audio *au=calloc(sizeof(monitor_audio),1);
+  strcpy(au->name,"Echo Reflector");
+  au->start=audio_reflector_null;
+  au->stop=audio_reflector_null;
+  au->poll_fds=audio_reflector_pollfds;
+  au->read=audio_reflector_read;
+  au->write=audio_reflector_write;
+  return au;
+}

audiodevices.c

@@ -61,15 +61,6 @@ int getAudioBytes(unsigned char *buffer,
   return -1;
 }
 
-/* as with recording, some of the devices have a fixed buffer size that
-   we must completely fill. 
-*/
-int playAudio(unsigned char *data,int bytes)
-{
-  if (audev&&audev->write) return audev->write(data,bytes);
-  return -1;
-}
-
 int stopAudio()
 {
   if (audev&&audev->stop) return audev->stop();

codecs.c

@@ -40,3 +40,14 @@ int encodeAndDispatchRecordedAudio(int fd,int callSessionToken,
   write(fd,msg,len);
   return 0;
 }
+
+int bufferAudioForPlayback(int codec,long long start_time,long long end_time,
+		unsigned char *data,int dataLen)
+{
+  /* XXX We need to buffer and reorder out-of-order sample blocks and
+     decode codecs etc here. */
+
+  /* send audio to device */
+  int bytesWritten=audev->write(&data[0],dataLen);
+  return 0;
+}

monitor-cli.c

@@ -48,6 +48,7 @@ int processChar(int c);
 
 int autoAnswerP=1;
 int pipeAudio=1;
+int reflectAudio=0;
 int syntheticAudio=0;
 int showReceived=1;
 int interactiveP=1;
@@ -62,6 +63,11 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
   cli_arg(argc, argv, o, "sid", &sid, NULL, "");
   struct sockaddr_un addr;
 
+  if (!strcasecmp(sid,"reflect")) {
+    pipeAudio=1; reflectAudio=1;
+    sid="";
+  }
+
   if ( (fd = socket(AF_UNIX, SOCK_STREAM, 0)) == -1) {
     perror("socket");
     exit(-1);
@@ -81,7 +87,10 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
   }
 
   if (pipeAudio) {
-    detectAudioDevice();
+    if (reflectAudio)
+      audev=audio_reflector_detect();
+    else 
+      detectAudioDevice();
     char *name=audev?audev->name:NULL;
     if (!name) {
       WHY("Could not detect any audio device. Will not pipe audio.");
@@ -118,17 +127,13 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
      we need, so we need to keep any left over ones from the previous
      read. */
   int audioRecordBufferBytes=0;
+  int audioRecordBufferSize=8000*2;
   unsigned char audioRecordBuffer[8000*2];
 
   int base_fd_count=fdcount;
   while(1) {
     fdcount=base_fd_count;
-    int audio_fd = getAudioRecordFd();
+    fdcount+=audev->poll_fds(&fds[fdcount],128-fdcount);
-    if (audio_fd>-1) {
-      fds[fdcount].fd=STDIN_FILENO;
-      fds[fdcount].events=POLLIN;
-      fdcount++;
-    }
     poll(fds,fdcount,1000);
 
     fcntl(fd,F_SETFL,
@@ -136,9 +141,6 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
     if (interactiveP) 
       fcntl(STDIN_FILENO,F_SETFL,
 	    fcntl(STDIN_FILENO, F_GETFL, NULL)|O_NONBLOCK);
-    if (audio_fd>-1) 
-      fcntl(audio_fd,F_SETFL,
-	    fcntl(audio_fd, F_GETFL, NULL)|O_NONBLOCK);
     
     int bytes;
     int i;
@@ -153,12 +155,14 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
 	printf("< %s",line);
 	write(fd,line,bytes);
       }
-      if (audio_fd>-1) {
+    }
-	/* attempt to read next audio buffer.
+
-	   Some audio devices that we are using require an entire buffer
+    if (audev&&audev->read)
-	   to be read at a time. */
+      {
-	audioRecordBufferBytes
+	int bytesRead=audev->read(&audioRecordBuffer[audioRecordBufferBytes],
-	  =getAudioBytes(audioRecordBuffer,audioRecordBufferBytes,sizeof(audioRecordBuffer));
+				  audioRecordBufferSize-audioRecordBufferBytes);
+	if (bytesRead>0) audioRecordBufferBytes+=bytesRead;
+	
 	/* 8KHz 16 bit samples = 16000 bytes per second.
 	   Thus one 1ms of audio = 16 bytes. */
 	int audioRecordBufferOffset=0;
@@ -180,13 +184,11 @@ int app_monitor_cli(int argc, const char *const *argv, struct command_line_optio
 	      audioRecordBufferBytes-audioRecordBufferOffset);
 	audioRecordBufferBytes-=audioRecordBufferOffset;
       }
-    }
+    
-
+    fcntl(fd,F_SETFL,
-  fcntl(fd,F_SETFL,
+	  fcntl(fd, F_GETFL, NULL)&~O_NONBLOCK);
-	fcntl(fd, F_GETFL, NULL)&~O_NONBLOCK);
+    fcntl(STDIN_FILENO,F_SETFL,
-  fcntl(STDIN_FILENO,F_SETFL,
+	  fcntl(STDIN_FILENO, F_GETFL, NULL)&~O_NONBLOCK);   
-	fcntl(STDIN_FILENO, F_GETFL, NULL)&~O_NONBLOCK);
-
   }
   
   return 0;
@@ -220,8 +222,8 @@ int processLine(char *cmd,unsigned char *data,int dataLen)
 	     &l_id,&r_id,&l_state,&r_state,
 	     &codec,&start_time,&end_time)==7)
     {
-      if (pipeAudio) {
+      if (pipeAudio&&audev) {
-	playAudio(data,dataLen);
+	bufferAudioForPlayback(codec,start_time,end_time,data,dataLen);	
       }
     }
   if (sscanf(cmd,"CALLSTATUS:%x:%x:%d:%d",
@@ -268,6 +270,7 @@ int processLine(char *cmd,unsigned char *data,int dataLen)
   state=STATE_CMD;
   return 0;
 }
+
 int processChar(int c)
 {
   switch(state) {

monitor.c

@@ -361,6 +361,7 @@ int monitor_process_command(int index,char *cmd)
   }
 
   char msg[1024];
+  int flag;
 
   if (cmd[0]=='*') {
     /* command with content */
@@ -391,6 +392,17 @@ int monitor_process_command(int index,char *cmd)
     c->flags|=MONITOR_RHIZOME;
   else if (!strcasecmp(cmd,"ignore rhizome"))
     c->flags&=~MONITOR_RHIZOME;
+  else if (sscanf(cmd,"FASTAUDIO:%x:%d",&callSessionToken,&flag)==2)
+    {
+      int i;
+      for(i=0;i<vomp_call_count;i++)
+	if (vomp_call_states[i].local.session==callSessionToken
+	    ||callSessionToken==0) {
+	  vomp_call_states[i].fast_audio=flag;
+	  vomp_call_states[i].local.last_state=-1;
+	  monitor_call_status(&vomp_call_states[i]);	  
+	}
+    }
   else if (sscanf(cmd,"call %s %s %s",sid,localDid,remoteDid)==3) {
     if (sid[0]=='*') {
       /* For testing, pick a peer and call them */
@@ -515,9 +527,10 @@ int monitor_call_status(vomp_call_state *call)
   call->remote.last_state=call->remote.state;
   if (show) {
     if (0) WHYF("sending call status to monitor");
-    snprintf(msg,1024,"\nCALLSTATUS:%06x:%06x:%d:%d:%s:%s:%s:%s\n",
+    snprintf(msg,1024,"\nCALLSTATUS:%06x:%06x:%d:%d:%d:%s:%s:%s:%s\n",
 	     call->local.session,call->remote.session,
 	     call->local.state,call->remote.state,
+	     call->fast_audio,
 	     overlay_render_sid(call->local.sid),
 	     overlay_render_sid(call->remote.sid),
 	     call->local.did,call->remote.did);

serval.h

@@ -1320,6 +1320,7 @@ typedef struct vomp_call_state {
   vomp_call_half local;
   vomp_call_half remote;
   int ringing;
+  int fast_audio;
   unsigned long long create_time;
   unsigned long long last_activity;
   unsigned long long audio_clock;
@@ -1444,17 +1445,19 @@ extern monitor_audio *audev;
 
 monitor_audio *audio_msm_g1_detect();
 monitor_audio *audio_alsa_detect();
+monitor_audio *audio_reflector_detect();
 int detectAudioDevice();
 int getAudioPlayFd();
 int getAudioRecordFd();
 int getAudioBytes(unsigned char *buffer,
 		  int offset,
 		  int bufferSize);
-int playAudio(unsigned char *data,int bytes);
-int startAudio();
-int stopAudio();
 int encodeAndDispatchRecordedAudio(int fd,int callSessionToken,
 				   int recordCodec,
 				   unsigned char *sampleData,
 				   int sampleBytes);
 int scrapeProcNetRoute();
+int bufferAudioForPlayback(int codec,long long start_time,long long end_time,
+			   unsigned char *data,int dataLen);
+int startAudio();
+int stopAudio();