Commit b0880792 authored by Simon Morlat's avatar Simon Morlat

add support for speex agc within MSVolume

enable it in linphonecore.
parent eb05a04d
......@@ -411,6 +411,8 @@ void sound_config_read(LinphoneCore *lc)
linphone_core_enable_echo_limiter(lc,
lp_config_get_int(lc->config,"sound","echolimiter",0));
linphone_core_enable_agc(lc,
lp_config_get_int(lc->config,"sound","agc",0));
}
void sip_config_read(LinphoneCore *lc)
......@@ -1435,6 +1437,7 @@ void linphone_core_init_media_streams(LinphoneCore *lc){
}
}
audio_stream_enable_automatic_gain_control(lc->audiostream,linphone_core_agc_enabled(lc));
#ifdef VIDEO_ENABLED
if (lc->video_conf.display || lc->video_conf.capture)
lc->videostream=video_stream_new(linphone_core_get_video_port(lc),linphone_core_ipv6_enabled(lc));
......@@ -1951,6 +1954,14 @@ bool_t linphone_core_echo_limiter_enabled(const LinphoneCore *lc){
return lc->sound_conf.ea;
}
void linphone_core_enable_agc(LinphoneCore *lc, bool_t val){
lc->sound_conf.agc=val;
}
bool_t linphone_core_agc_enabled(const LinphoneCore *lc){
return lc->sound_conf.agc;
}
void linphone_core_send_dtmf(LinphoneCore *lc,char dtmf)
{
......
......@@ -107,6 +107,7 @@ typedef struct sound_config
char *remote_ring;
bool_t ec;
bool_t ea;
bool_t agc;
} sound_config_t;
typedef struct codecs_config
......@@ -713,6 +714,9 @@ bool_t linphone_core_echo_cancelation_enabled(LinphoneCore *lc);
void linphone_core_enable_echo_limiter(LinphoneCore *lc, bool_t val);
bool_t linphone_core_echo_limiter_enabled(const LinphoneCore *lc);
void linphone_core_enable_agc(LinphoneCore *lc, bool_t val);
bool_t linphone_core_agc_enabled(const LinphoneCore *lc);
void linphone_core_set_presence_info(LinphoneCore *lc,int minutes_away,const char *contact,LinphoneOnlineStatus os);
LinphoneOnlineStatus linphone_core_get_presence_info(const LinphoneCore *lc);
......
......@@ -8,10 +8,24 @@
<primarylanguage>C</primarylanguage>
<ignoreparts/>
<projectname>linphone</projectname>
<projectdirectory>.</projectdirectory>
<absoluteprojectpath>false</absoluteprojectpath>
<description></description>
<defaultencoding></defaultencoding>
</general>
<kdevcustomproject>
<run>
<directoryradio>executable</directoryradio>
<mainprogram>/home/smorlat/sources/git/linphone/linphone</mainprogram>
<programargs></programargs>
<globaldebugarguments></globaldebugarguments>
<globalcwd>/home/smorlat/sources/git/linphone/linphone</globalcwd>
<useglobalprogram>false</useglobalprogram>
<terminal>false</terminal>
<autocompile>false</autocompile>
<autoinstall>false</autoinstall>
<autokdesu>false</autokdesu>
<envvars/>
</run>
<filetypes>
<filetype>*.java</filetype>
......@@ -374,11 +388,51 @@
<path>win32acm/wineacm.h</path>
<path>win32acm/wrapper.h</path>
</blacklist>
<build>
<buildtool>make</buildtool>
<builddir></builddir>
</build>
<other>
<prio>0</prio>
<otherbin></otherbin>
<defaulttarget></defaulttarget>
<otheroptions></otheroptions>
<selectedenvironment>default</selectedenvironment>
<environments>
<default/>
</environments>
</other>
<make>
<abortonerror>true</abortonerror>
<numberofjobs>0</numberofjobs>
<prio>0</prio>
<dontact>false</dontact>
<makebin></makebin>
<defaulttarget></defaulttarget>
<makeoptions></makeoptions>
<selectedenvironment>default</selectedenvironment>
<environments>
<default/>
</environments>
</make>
</kdevcustomproject>
<kdevdebugger>
<general>
<dbgshell/>
<dbgshell></dbgshell>
<gdbpath></gdbpath>
<configGdbScript></configGdbScript>
<runShellScript></runShellScript>
<runGdbScript></runGdbScript>
<breakonloadinglibs>true</breakonloadinglibs>
<separatetty>false</separatetty>
<floatingtoolbar>false</floatingtoolbar>
<raiseGDBOnStart>false</raiseGDBOnStart>
</general>
<display>
<staticmembers>false</staticmembers>
<demanglenames>true</demanglenames>
<outputradix>10</outputradix>
</display>
</kdevdebugger>
<kdevdoctreeview>
<ignoretocs>
......@@ -470,6 +524,19 @@
<alwaysIncludeNamespaces>false</alwaysIncludeNamespaces>
<includePaths>.;</includePaths>
</codecompletion>
<creategettersetter>
<prefixGet></prefixGet>
<prefixSet>set</prefixSet>
<prefixVariable>m_,_</prefixVariable>
<parameterName>theValue</parameterName>
<inlineGet>true</inlineGet>
<inlineSet>true</inlineSet>
</creategettersetter>
<splitheadersource>
<enabled>false</enabled>
<synchronize>true</synchronize>
<orientation>Vertical</orientation>
</splitheadersource>
</kdevcppsupport>
<kdevfileview>
<groups>
......@@ -481,4 +548,10 @@
<hidenonprojectfiles>false</hidenonprojectfiles>
</tree>
</kdevfileview>
<cppsupportpart>
<filetemplates>
<interfacesuffix>.h</interfacesuffix>
<implementationsuffix>.cpp</implementationsuffix>
</filetemplates>
</cppsupportpart>
</kdevelop>
......@@ -53,6 +53,7 @@ struct _AudioStream
EchoLimiterType el_type; /*use echo limiter: two MSVolume, measured input level controlling local output level*/
bool_t play_dtmfs;
bool_t use_gc;
bool_t use_agc;
};
#ifdef __cplusplus
......@@ -105,6 +106,9 @@ void audio_stream_enable_echo_limiter(AudioStream *stream, EchoLimiterType type)
/*enable gain control, to be done before start() */
void audio_stream_enable_gain_control(AudioStream *stream, bool_t val);
/*enable automatic gain control, to be done before start() */
void audio_stream_enable_automatic_gain_control(AudioStream *stream, bool_t val);
void audio_stream_set_mic_gain(AudioStream *stream, float gain);
/* stop the audio streaming thread and free everything*/
......
......@@ -48,6 +48,8 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
#define MS_VOLUME_SET_EA_FORCE MS_FILTER_METHOD(MS_VOLUME_ID,7,float)
#define MS_VOLUME_ENABLE_AGC MS_FILTER_METHOD(MS_VOLUME_ID,8,int)
extern MSFilterDesc ms_volume_desc;
#endif
......@@ -257,6 +257,13 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
}
}
if (stream->use_agc){
int tmp=1;
if (stream->volsend==NULL)
stream->volsend=ms_filter_new(MS_VOLUME_ID);
ms_filter_call_method(stream->volsend,MS_VOLUME_ENABLE_AGC,&tmp);
}
/* give the sound filters some properties */
ms_filter_call_method(stream->soundread,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
ms_filter_call_method(stream->soundwrite,MS_FILTER_SET_SAMPLE_RATE,&pt->clock_rate);
......@@ -382,6 +389,7 @@ AudioStream *audio_stream_new(int locport, bool_t ipv6){
stream->rtpsend=ms_filter_new(MS_RTP_SEND_ID);
stream->play_dtmfs=TRUE;
stream->use_gc=FALSE;
stream->use_agc=FALSE;
return stream;
}
......@@ -406,6 +414,10 @@ void audio_stream_enable_gain_control(AudioStream *stream, bool_t val){
stream->use_gc=val;
}
void audio_stream_enable_automatic_gain_control(AudioStream *stream, bool_t val){
stream->use_agc=val;
}
void audio_stream_set_mic_gain(AudioStream *stream, float gain){
if (stream->volsend){
ms_filter_call_method(stream->volsend,MS_VOLUME_SET_GAIN,&gain);
......
......@@ -17,9 +17,17 @@ along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "mediastreamer-config.h"
#endif
#include "mediastreamer2/msvolume.h"
#include <math.h>
#ifdef HAVE_SPEEXDSP
#include <speex/speex_preprocess.h>
#endif
static const float max_e=32767*32767;
static const float coef=0.1;
static const float gain_k=0.02;
......@@ -36,7 +44,14 @@ typedef struct Volume{
float thres;
float force;
MSFilter *peer;
#ifdef HAVE_SPEEXDSP
SpeexPreprocessState *speex_pp;
#endif
int sample_rate;
int nsamples;
MSBufferizer *buffer;
bool_t ea_active;
bool_t agc_enabled;
}Volume;
static void volume_init(MSFilter *f){
......@@ -49,10 +64,19 @@ static void volume_init(MSFilter *f){
v->thres=noise_thres;
v->force=en_weight;
v->peer=NULL;
v->agc_enabled=FALSE;
v->buffer=ms_bufferizer_new();
v->sample_rate=8000;
v->nsamples=80;
#ifdef HAVE_SPEEXDSP
v->speex_pp=NULL;
#endif
f->data=v;
}
static void volume_uninit(MSFilter *f){
Volume *v=(Volume*)f->data;
ms_bufferizer_destroy(v->buffer);
ms_free(f->data);
}
......@@ -63,12 +87,29 @@ static int volume_get(MSFilter *f, void *arg){
return 0;
}
static int volume_set_sample_rate(MSFilter *f, void *arg){
Volume *v=(Volume*)f->data;
v->sample_rate=*(int*)arg;
return 0;
}
static int volume_get_linear(MSFilter *f, void *arg){
float *farg=(float*)arg;
Volume *v=(Volume*)f->data;
*farg=(v->energy+1)/max_e;
return 0;
}
#ifdef HAVE_SPEEXDSP
static void volume_agc_process(Volume *v, mblk_t *om){
speex_preprocess_run(v->speex_pp,(int16_t*)om->b_rptr);
}
#else
static void volume_agc_process(Volume *v, mblk_t *om){
}
#endif
static inline float compute_gain(float static_gain, float energy, float weight){
float ret=static_gain*(1 - (energy*weight));
......@@ -129,6 +170,12 @@ static int volume_set_peer(MSFilter *f, void *arg){
return 0;
}
static int volume_set_agc(MSFilter *f, void *arg){
Volume *v=(Volume*)f->data;
v->agc_enabled=*(int*)arg;
return 0;
}
static int volume_set_ea_threshold(MSFilter *f, void*arg){
Volume *v=(Volume*)f->data;
float val=*(float*)arg;
......@@ -162,30 +209,88 @@ static inline int16_t saturate(float val){
return (val>32767) ? 32767 : ( (val<-32767) ? -32767 : val);
}
static float update_energy(int16_t *signal, int numsamples, float last_energy_value){
int i;
float en=last_energy_value;
for (i=0;i<numsamples;++i){
float s=(float)signal[i];
en=(s*s*coef) + (1.0-coef)*en;
}
return en;
}
static void apply_gain(mblk_t *m, float gain){
int16_t *sample;
for ( sample=(int16_t*)m->b_rptr;
sample<(int16_t*)m->b_wptr;
++sample){
float s=*sample;
*sample=saturate(s*gain);
}
}
static void volume_preprocess(MSFilter *f){
Volume *v=(Volume*)f->data;
/*process agc by chunks of 10 ms*/
v->nsamples=(int)(0.01*(float)v->sample_rate);
if (v->agc_enabled){
ms_message("AGC is enabled.");
#ifdef HAVE_SPEEXDSP
if (v->speex_pp==NULL){
int tmp=1;
v->speex_pp=speex_preprocess_state_init(v->nsamples,v->sample_rate);
if (speex_preprocess_ctl(v->speex_pp,SPEEX_PREPROCESS_SET_AGC,&tmp)==-1){
ms_warning("Speex AGC is not available.");
}
tmp=0;
speex_preprocess_ctl(v->speex_pp,SPEEX_PREPROCESS_SET_VAD,&tmp);
speex_preprocess_ctl(v->speex_pp,SPEEX_PREPROCESS_SET_DENOISE,&tmp);
speex_preprocess_ctl(v->speex_pp,SPEEX_PREPROCESS_SET_DEREVERB,&tmp);
}
#else
ms_error("No AGC possible, mediastreamer2 was compiled without libspeexdsp.");
#endif
}
}
static void volume_process(MSFilter *f){
mblk_t *m;
int16_t *sample;
Volume *v=(Volume*)f->data;
float en=v->energy;
while((m=ms_queue_get(f->inputs[0]))!=NULL){
for ( sample=(int16_t*)m->b_rptr;
sample<(int16_t*)m->b_wptr;
++sample){
float s=*sample;
en=(s*s*coef) + (1.0-coef)*en;
}
if (v->peer){
volume_echo_avoider_process(v);
if (v->agc_enabled){
mblk_t *om;
int nbytes=v->nsamples*2;
ms_bufferizer_put_from_queue(v->buffer,f->inputs[0]);
while(ms_bufferizer_get_avail(v->buffer)>=nbytes){
om=allocb(nbytes,0);
ms_bufferizer_read(v->buffer,om->b_wptr,nbytes);
om->b_wptr+=nbytes;
en=update_energy((int16_t*)om->b_rptr,om->b_wptr-om->b_rptr,en);
volume_agc_process(v,om);
if (v->peer){
volume_echo_avoider_process(v);
}
if (v->gain!=1){
apply_gain(om,v->gain);
}
ms_queue_put(f->outputs[0],om);
}
if (v->gain!=1){
for ( sample=(int16_t*)m->b_rptr;
sample<(int16_t*)m->b_wptr;
++sample){
float s=*sample;
*sample=saturate(s*v->gain);
}else{
/*light processing: no agc. Work in place in the input buffer*/
while((m=ms_queue_get(f->inputs[0]))!=NULL){
en=update_energy((int16_t*)m->b_rptr,m->b_wptr-m->b_rptr,en);
if (v->peer){
volume_echo_avoider_process(v);
}
if (v->gain!=1){
apply_gain(m,v->gain);
}
ms_queue_put(f->outputs[0],m);
}
ms_queue_put(f->outputs[0],m);
}
v->energy=en;
}
......@@ -199,19 +304,22 @@ static MSFilterMethod methods[]={
{ MS_VOLUME_SET_EA_THRESHOLD , volume_set_ea_threshold },
{ MS_VOLUME_SET_EA_SPEED , volume_set_ea_speed },
{ MS_VOLUME_SET_EA_FORCE , volume_set_ea_force },
{ MS_FILTER_SET_SAMPLE_RATE, volume_set_sample_rate },
{ MS_VOLUME_ENABLE_AGC , volume_set_agc },
{ 0 , NULL }
};
#ifndef _MSC_VER
MSFilterDesc ms_volume_desc={
.name="MSVolume",
.text=N_("A filter to make level measurements on 16 bits pcm audio stream"),
.text=N_("A filter that controls and measure sound volume"),
.id=MS_VOLUME_ID,
.category=MS_FILTER_OTHER,
.ninputs=1,
.noutputs=1,
.init=volume_init,
.uninit=volume_uninit,
.preprocess=volume_preprocess,
.process=volume_process,
.methods=methods
};
......@@ -219,7 +327,7 @@ MSFilterDesc ms_volume_desc={
MSFilterDesc ms_volume_desc={
MS_VOLUME_ID,
"MSVolume",
N_("A filter to make level measurements on 16 bits pcm audio stream"),
N_("A filter that controls and measure sound volume"),
MS_FILTER_OTHER,
NULL,
1,
......
......@@ -122,8 +122,10 @@ const char *usage="mediastream --local <port> --remote <ip:port> --payload <payl
"[ --jitter <miliseconds>]\n"
"[ --width <pixels>]\n"
"[ --height <pixels> ]\n"
"[ --bitrate <bits per seconds>]\n";
static void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp, int jitter, bool_t ec, int bitrate, MSVideoSize vs);
"[ --bitrate <bits per seconds>]\n"
"[ --ec (enable echo canceller)]\n"
"[ --agc (enable automatic gain control)]\n";
static void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp, int jitter, bool_t ec, int bitrate, MSVideoSize vs, bool_t agc);
int main(int argc, char * argv[])
......@@ -136,6 +138,7 @@ int main(int argc, char * argv[])
int bitrate=0;
MSVideoSize vs;
bool_t ec=FALSE;
bool_t agc=FALSE;
/*create the rtp session */
ortp_init();
ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
......@@ -188,14 +191,16 @@ int main(int argc, char * argv[])
vs.height=atoi(argv[i]);
}else if (strcmp(argv[i],"--ec")==0){
ec=TRUE;
}else if (strcmp(argv[i],"--agc")==0){
agc=TRUE;
}
}
run_media_streams(localport,ip,remoteport,payload,fmtp,jitter,ec,bitrate,vs);
run_media_streams(localport,ip,remoteport,payload,fmtp,jitter,ec,bitrate,vs, agc);
return 0;
}
void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp, int jitter, bool_t ec, int bitrate, MSVideoSize vs)
void run_media_streams(int localport, const char *remote_ip, int remoteport, int payload, const char *fmtp, int jitter, bool_t ec, int bitrate, MSVideoSize vs, bool_t agc)
{
AudioStream *audio=NULL;
#ifdef VIDEO_ENABLED
......@@ -218,7 +223,13 @@ void run_media_streams(int localport, const char *remote_ip, int remoteport, in
if (pt->type!=PAYLOAD_VIDEO){
printf("Starting audio stream.\n");
audio=audio_stream_start(profile,localport,remote_ip,remoteport,payload,jitter, ec);
MSSndCardManager *manager=ms_snd_card_manager_get();
audio=audio_stream_new(localport,ms_is_ipv6(remote_ip));
audio_stream_enable_automatic_gain_control(audio,agc);
audio_stream_start_now(audio,profile,remote_ip,remoteport,remoteport+1,payload,jitter,
ms_snd_card_manager_get_default_playback_card(manager),
ms_snd_card_manager_get_default_capture_card(manager),
ec);
if (audio) session=audio->session;
}else{
#ifdef VIDEO_ENABLED
......
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