Fix compilation warnings when building for Windows 10.

include/mediastreamer2/msqueue.h

@@ -113,7 +113,7 @@ MS2_PUBLIC void ms_queue_destroy(MSQueue *q);
 	
 struct _MSBufferizer{
 	queue_t q;
-	int size;
+	size_t size;
 };
 
 typedef struct _MSBufferizer MSBufferizer;
@@ -136,7 +136,7 @@ MS2_PUBLIC int ms_bufferizer_read(MSBufferizer *obj, uint8_t *data, int datalen)
 MS2_PUBLIC void ms_bufferizer_fill_current_metas(MSBufferizer *obj, mblk_t *m);
 
 /* returns the number of bytes available in the bufferizer*/
-static MS2_INLINE int ms_bufferizer_get_avail(MSBufferizer *obj){
+static MS2_INLINE size_t ms_bufferizer_get_avail(MSBufferizer *obj){
 	return obj->size;
 }
 

src/audiofilters/audiomixer.c

@@ -97,7 +97,7 @@ static int channel_flow_control(Channel *chan, int threshold, uint64_t time){
 		chan->min_fullness=-1;
 		return skip;
 	}
-	size=ms_bufferizer_get_avail(&chan->bufferizer);
+	size=(int)ms_bufferizer_get_avail(&chan->bufferizer);
 	if (chan->min_fullness==-1 || chan->min_fullness<size) chan->min_fullness=size;
 	if (time-chan->last_flow_control>=5000){
 		if (chan->min_fullness>=threshold){

src/audiofilters/chanadapt.c

@@ -43,7 +43,7 @@ static void adapter_uninit(MSFilter *f){
 static void adapter_process(MSFilter *f){
 	AdapterState *s=(AdapterState*)f->data;
 	mblk_t *im,*om;
-	int msgsize;
+	size_t msgsize;
 	
 	while((im=ms_queue_get(f->inputs[0]))!=NULL){
 		if (s->inputchans==s->outputchans){

src/audiofilters/dtmfgen.c

@@ -63,8 +63,8 @@ static void dtmfgen_init(MSFilter *f){
 	s->pos=0;
 	s->nosamples_time=0;
 	s->silence=0;
-	s->default_amplitude=0.2;
-	s->amplitude=(s->default_amplitude*0.7*32767);
+	s->default_amplitude=0.2f;
+	s->amplitude=(int)(s->default_amplitude*0.7*32767);
 	s->repeat_count=0;
 	f->data=s;
 }
@@ -159,7 +159,7 @@ static int dtmfgen_put(MSFilter *f, void *arg){
 	s->highfreq=s->highfreq/s->rate;
 	s->dur=s->rate/10; /*100 ms duration */
 	s->silence=0;
-	s->amplitude=s->default_amplitude*32767*0.7;
+	s->amplitude=(int)(s->default_amplitude*32767*0.7);
 	s->current_tone.tone_name[0]=dtmf[0];
 	s->current_tone.tone_name[1]=0;
 	s->current_tone.interval=0;
@@ -181,7 +181,7 @@ static int dtmfgen_play_tone(MSFilter *f, void *arg){
 	s->lowfreq=((float)def->frequencies[0])/(float)s->rate;
 	s->highfreq=((float)def->frequencies[1])/(float)s->rate;;
 	s->silence=0;
-	s->amplitude=((float)def->amplitude)* 0.7*32767.0;
+	s->amplitude=(int)(def->amplitude* 0.7*32767.0);
 	s->repeat_count=0;
 	s->playing=TRUE;
 	ms_filter_unlock(f);
@@ -314,7 +314,7 @@ static void dtmfgen_process(MSFilter *f){
 					strncpy(ev.tone_name,s->current_tone.tone_name,sizeof(ev.tone_name));
 					ms_filter_notify(f,MS_DTMF_GEN_EVENT,&ev);
 				}
-				nsamples=(m->b_wptr-m->b_rptr)/(2*s->nchannels);
+				nsamples=(int)(m->b_wptr-m->b_rptr)/(2*s->nchannels);
 				write_dtmf(s, (int16_t*)m->b_rptr,nsamples);
 			}
 			ms_queue_put(f->outputs[0],m);

src/audiofilters/equalizer.c

@@ -29,7 +29,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #ifdef MS_FIXED_POINT
 #define GAIN_ZERODB 20000
 #else
-#define GAIN_ZERODB 1.0
+#define GAIN_ZERODB 1.0f
 #endif
 
 #define EQUALIZER_DEFAULT_RATE 8000
@@ -47,7 +47,7 @@ typedef struct _EqualizerState{
 
 static void equalizer_state_flatten(EqualizerState *s){
 	int i;
-	ms_word16_t val=GAIN_ZERODB/s->nfft;
+	ms_word16_t val=(ms_word16_t)(GAIN_ZERODB/s->nfft);
 	s->fft_cpx[0]=val;
 	for(i=1;i<s->nfft;i+=2)
 		s->fft_cpx[i]=val;
@@ -150,7 +150,7 @@ static void equalizer_state_set(EqualizerState *s, int freq_0, float gain, int f
 	//int low,high;
 	int i, f;
 	int delta_f = equalizer_state_index2hz(s, 1);
-	float sqrt_gain = sqrt(gain);
+	float sqrt_gain = (float)sqrt(gain);
 	int mid = equalizer_state_hz_to_index(s, freq_0);
 	freq_bw-= delta_f/2;   /* subtract a constant - compensates for limited fft steepness at low f */
 	if (freq_bw < delta_f/2)
@@ -210,8 +210,8 @@ static void norm_and_apodize(ms_word16_t *s, int len){
 	float x;
 	float w;
 	for(i=0;i<len;++i){
-		x=(float)i*2*M_PI/(float)len;
-		w=0.54 - (0.46*cos(x));
+		x=(float)((float)i*2*M_PI/(float)len);
+		w=(float)(0.54 - (0.46*cos(x)));
 		//w=0.42 - (0.5*cos(x)) + (0.08*cos(2*x));
 		s[i]=w*(float)s[i];
 	}
@@ -285,7 +285,7 @@ static void equalizer_process(MSFilter *f){
 	EqualizerState *s=(EqualizerState*)f->data;
 	while((m=ms_queue_get(f->inputs[0]))!=NULL){
 		if (s->active){
-			equalizer_state_run(s,(int16_t*)m->b_rptr,(m->b_wptr-m->b_rptr)/2);
+			equalizer_state_run(s,(int16_t*)m->b_rptr,(int)((m->b_wptr-m->b_rptr)/2));
 		}
 		ms_queue_put(f->outputs[0],m);
 	}
@@ -294,14 +294,14 @@ static void equalizer_process(MSFilter *f){
 static int equalizer_set_gain(MSFilter *f, void *data){
 	EqualizerState *s=(EqualizerState*)f->data;
 	MSEqualizerGain *d=(MSEqualizerGain*)data;
-	equalizer_state_set(s,d->frequency,d->gain,d->width);
+	equalizer_state_set(s,(int)d->frequency,d->gain,(int)d->width);
 	return 0;
 }
 
 static int equalizer_get_gain(MSFilter *f, void *data){
 	EqualizerState *s=(EqualizerState*)f->data;
 	MSEqualizerGain *d=(MSEqualizerGain*)data;
-	d->gain=equalizer_state_get(s,d->frequency);
+	d->gain=equalizer_state_get(s,(int)d->frequency);
 	d->width=0;
 	return 0;
 }

src/audiofilters/flowcontrol.c

@@ -75,7 +75,7 @@ static void discard_well_choosed_samples(mblk_t *m, int nsamples, int todrop)
 
 mblk_t *ms_audio_flow_controller_process(MSAudioFlowController *ctl, mblk_t *m){
 	if (ctl->total_samples > 0 && ctl->target_samples > 0) {
-		int nsamples = (m->b_wptr - m->b_rptr) / 2;
+		int nsamples = (int)((m->b_wptr - m->b_rptr) / 2);
 		int th_dropped;
 		int todrop;
 

src/audiofilters/genericplc.c

@@ -75,7 +75,7 @@ static void generic_plc_process(MSFilter *f) {
 	while((m=ms_queue_get(f->inputs[0]))!=NULL){
 		int transitionBufferSize = mgps->rate*sizeof(int16_t)*TRANSITION_DELAY/1000;
 		unsigned char buf[128];
-		unsigned int time = (1000*(m->b_wptr - m->b_rptr))/(mgps->rate*sizeof(int16_t)*mgps->nchannels);
+		unsigned int time = (unsigned int)((1000*(m->b_wptr - m->b_rptr))/(mgps->rate*sizeof(int16_t)*mgps->nchannels));
 		ms_concealer_inc_sample_time(mgps->concealer, f->ticker->time, time, TRUE);
 
 		/* introduce 16 sample delay (2ms) */

src/audiofilters/gsm.c

@@ -93,7 +93,7 @@ static void enc_process(MSFilter *f){
 	unsigned int unitary_buff_size = sizeof(int16_t)*160;
 	unsigned int buff_size = unitary_buff_size*s->ptime/20;
 	int16_t* buff;
-	int offset;
+	unsigned int offset;
 	
 	while((im=ms_queue_get(f->inputs[0]))!=NULL){
 		ms_bufferizer_put(s->bufferizer,im);

src/audiofilters/l16.c

@@ -204,7 +204,7 @@ static void dec_process(MSFilter *f)
 	mblk_t *im;
 
 	while((im=ms_queue_get(f->inputs[0]))) {
-		network_to_host((int16_t*)im->b_rptr,(im->b_wptr-im->b_rptr)/2);
+		network_to_host((int16_t*)im->b_rptr,(int)((im->b_wptr-im->b_rptr)/2));
 		ms_queue_put(f->outputs[0],im);
 	}
 };

src/audiofilters/msfilerec.c

@@ -54,7 +54,7 @@ static void rec_process(MSFilter *f){
 		ms_mutex_lock(&f->lock);
 		if (s->state==MSRecorderRunning){
 			while(it!=NULL){
-				int len=it->b_wptr-it->b_rptr;
+				int len=(int)(it->b_wptr-it->b_rptr);
 				if ((err=write(s->fd,it->b_rptr,len))!=len){
 					if (err<0)
 						ms_warning("MSFileRec: fail to write %i bytes: %s",len,strerror(errno));

src/audiofilters/msg722.c

@@ -206,7 +206,7 @@ static void dec_process(MSFilter *f)
 	mblk_t *om;
 
 	while((im=ms_queue_get(f->inputs[0]))) {
-		int payloadlen = im->b_wptr - im->b_rptr;
+		int payloadlen = (int)(im->b_wptr - im->b_rptr);
 		int declen;
 
 		om=allocb(payloadlen*4,0);

src/audiofilters/msopus.c

@@ -303,7 +303,7 @@ static void compute_max_bitrate(OpusEncData *d, int ptimeStep) {
 	if (normalized_cbr<6000) {
 		int initial_value = normalized_cbr;
 		normalized_cbr = 6000;
-		d->max_network_bitrate = (normalized_cbr/(pps*8) + 12 + 8 + 20) *8*pps;
+		d->max_network_bitrate = (int)((normalized_cbr/(pps*8) + 12 + 8 + 20) *8*pps);
 		ms_warning("Opus encoder does not support bitrate [%i]. Instead set to 6kbps, network bitrate [%d]", initial_value, d->max_network_bitrate);
 	}
 	if (d->maxaveragebitrate>0)
@@ -312,7 +312,7 @@ static void compute_max_bitrate(OpusEncData *d, int ptimeStep) {
 	if (normalized_cbr>maxaveragebitrate) {
 		int initial_value = normalized_cbr;
 		normalized_cbr = maxaveragebitrate;
-		d->max_network_bitrate = (normalized_cbr/(pps*8) + 12 + 8 + 20) *8*pps;
+		d->max_network_bitrate = (int)((normalized_cbr/(pps*8) + 12 + 8 + 20) *8*pps);
 		ms_warning("Opus encoder cannot set codec bitrate to [%i] because of maxaveragebitrate constraint or absolute maximum bitrate value. New network bitrate is [%i]", initial_value, d->max_network_bitrate);
 	}
 	ms_message("MSOpusEnc: codec bitrate set to [%i] with ptime [%i]", normalized_cbr, d->ptime);
@@ -694,7 +694,7 @@ static void ms_opus_dec_process(MSFilter *f) {
 	while ((im = ms_queue_get(f->inputs[0])) != NULL) {
 		om = allocb(5760 * d->channels * SIGNAL_SAMPLE_SIZE, 0); /* 5760 is the maximum number of sample in a packet (120ms at 48KHz) */
 
-		frames = opus_decode(d->state, (const unsigned char *)im->b_rptr, im->b_wptr - im->b_rptr, (opus_int16 *)om->b_wptr, 5760, 0);
+		frames = opus_decode(d->state, (const unsigned char *)im->b_rptr, (opus_int32)(im->b_wptr - im->b_rptr), (opus_int16 *)om->b_wptr, 5760, 0);
 
 		if (frames < 0) {
 			ms_warning("Opus decoder error: %s", opus_strerror(frames));

src/audiofilters/msresample.c

@@ -87,14 +87,14 @@ static void resample_uninit(MSFilter *obj){
 
 static int resample_channel_adapt(int in_nchannels, int out_nchannels, mblk_t *im, mblk_t **om) {
 	if ((in_nchannels == 2) && (out_nchannels == 1)) {
-		int msgsize = msgdsize(im) / 2;
+		size_t msgsize = msgdsize(im) / 2;
 		*om = allocb(msgsize, 0);
 		for (; im->b_rptr < im->b_wptr; im->b_rptr += 4, (*om)->b_wptr += 2) {
 			*(int16_t *)(*om)->b_wptr = *(int16_t *)im->b_rptr;
 		}
 		return 1;
 	} else if ((in_nchannels == 1) && (out_nchannels == 2)) {
-		int msgsize = msgdsize(im) * 2;
+		size_t msgsize = msgdsize(im) * 2;
 		*om = allocb(msgsize, 0);
 		for (; im->b_rptr < im->b_wptr; im->b_rptr += 2, (*om)->b_wptr += 4) {
 			((int16_t *)(*om)->b_wptr)[0] = *(int16_t *)im->b_rptr;
@@ -156,7 +156,7 @@ static void resample_process_ms2(MSFilter *obj){
 
 	
 	while((im=ms_queue_get(obj->inputs[0]))!=NULL){
-		unsigned int inlen=(im->b_wptr-im->b_rptr)/(2*dt->in_nchannels);
+		unsigned int inlen=(int)((im->b_wptr-im->b_rptr)/(2*dt->in_nchannels));
 		unsigned int outlen=((inlen*dt->output_rate)/dt->input_rate)+1;
 		unsigned int inlen_orig=inlen;
 		om=allocb(outlen*2*dt->in_nchannels,0);

src/audiofilters/msspeex.c

@@ -572,14 +572,14 @@ static void dec_process(MSFilter *f){
 	bool_t bits_initd=FALSE;
 
 	while((im=ms_queue_get(f->inputs[0]))!=NULL){
-		int rem_bits=(im->b_wptr-im->b_rptr)*8;
+		int rem_bits=(int)((im->b_wptr-im->b_rptr)*8);
 
 		if (!bits_initd) {
 			speex_bits_init(&bits);
 			bits_initd=TRUE;
 		}else speex_bits_reset(&bits);
 
-		speex_bits_read_from(&bits,(char*)im->b_rptr,im->b_wptr-im->b_rptr);
+		speex_bits_read_from(&bits,(char*)im->b_rptr,(int)(im->b_wptr-im->b_rptr));
 
 		/* support for multiple frame  in one RTP packet */
  		do{

src/audiofilters/msvaddtx.c

@@ -32,9 +32,9 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 
 #include <math.h>
 
-static const float max_e = (32768* 0.7);   /* 0.7 - is RMS factor */
-static const float coef = 0.2; /* floating averaging coeff. for energy */
-static const float silence_threshold=0.01;
+static const float max_e = (32768* 0.7f);   /* 0.7 - is RMS factor */
+static const float coef = 0.2f; /* floating averaging coeff. for energy */
+static const float silence_threshold=0.01f;
 
 typedef struct _VadDtxContext{
 	int silence_mode;/*set to 1 if a silence period is running*/
@@ -78,8 +78,8 @@ static void update_energy(VadDtxContext *v, int16_t *signal, int numsamples, uin
 		int s=signal[i];
 		acc += s * s;
 	}
-	en = (sqrt(acc / numsamples)+1) / max_e;
-	v->energy = (en * coef) + v->energy * (1.0 - coef);
+	en = (float)((sqrt(acc / numsamples)+1) / max_e);
+	v->energy = (en * coef) + v->energy * (1.0f - coef);
 	ortp_extremum_record_max(&v->max,curtime,v->energy);
 	//ms_message("Energy=%f, current max=%f",v->energy, ortp_extremum_get_current(&v->max));
 }
@@ -127,7 +127,7 @@ static void vad_dtx_process(MSFilter *f){
 
 #else
 	while((m=ms_queue_get(f->inputs[0]))!=NULL){
-		update_energy(ctx,(int16_t*)m->b_rptr, (m->b_wptr - m->b_rptr) / 2, f->ticker->time);
+		update_energy(ctx,(int16_t*)m->b_rptr, (int)((m->b_wptr - m->b_rptr) / 2), f->ticker->time);
 
 		if (ortp_extremum_get_current(&ctx->max)<silence_threshold){
 			if (!ctx->silence_mode){

src/audiofilters/msvolume.c

@@ -30,15 +30,15 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #include <speex/speex_preprocess.h>
 #endif
 
-static const float max_e = (32768* 0.7);   /* 0.7 - is RMS factor */
-static const float coef = 0.2; /* floating averaging coeff. for energy */
+static const float max_e = (32768* 0.7f);   /* 0.7 - is RMS factor */
+static const float coef = 0.2f; /* floating averaging coeff. for energy */
 //static const float gain_k = 0.02; /* floating averaging coeff. for gain */
-static const float vol_upramp = 0.4;
-static const float vol_downramp = 0.4;   /* not yet runtime parameterizable */
+static const float vol_upramp = 0.4f;
+static const float vol_downramp = 0.4f;   /* not yet runtime parameterizable */
 static const float en_weight=4.0;
-static const float noise_thres=0.1;
+static const float noise_thres=0.1f;
 static const float transmit_thres=4;
-static const float min_ng_floorgain=0.005;
+static const float min_ng_floorgain=0.005f;
 static const float agc_threshold=0.5;
 
 
@@ -199,14 +199,14 @@ static MS2_INLINE float compute_gain(Volume *v, float energy, float weight) {
 static float volume_echo_avoider_process(Volume *v, mblk_t *om) {
 	static int counter;
 	float peer_e,peer_pk;
-	int nsamples = ((om->b_wptr - om->b_rptr) / 2);
+	int nsamples = (int)((om->b_wptr - om->b_rptr) / 2);
 	float mic_spk_ratio;
 	peer_e = ((Volume *)(v->peer->data))->energy;
 	peer_pk=((Volume *)(v->peer->data))->energy;
 
 	if (peer_pk>v->lt_speaker_en)
 		v->lt_speaker_en=peer_pk;
-	else v->lt_speaker_en=(0.005*peer_pk)+(0.995*v->lt_speaker_en);
+	else v->lt_speaker_en=(0.005f*peer_pk)+(0.995f*v->lt_speaker_en);
 	mic_spk_ratio=(v->energy/(v->lt_speaker_en+v->ea_thres));
 
 	/* where v->target_gain is not set, it is kept steady - not to modify elsewhere! */
@@ -239,7 +239,7 @@ static float volume_echo_avoider_process(Volume *v, mblk_t *om) {
 static void volume_noise_gate_process(Volume *v , float energy, mblk_t *om){
 	static int counter;
 	float tgain = v->ng_floorgain;  /* start with floorgain */
-	int nsamples=((om->b_wptr-om->b_rptr)/2);
+	int nsamples=(int)((om->b_wptr-om->b_rptr)/2);
 	if (energy > v->ng_threshold) {
 		v->ng_noise_dur = v->ng_cut_time;
 		tgain = 1.0;
@@ -252,7 +252,7 @@ static void volume_noise_gate_process(Volume *v , float energy, mblk_t *om){
 	}
 	/* simple averaging computation is adequate here: fast rise, relatively slower decrease */
 	/* of gain - ears impression */
-	v->ng_gain = v->ng_gain*0.75 + tgain*0.25;
+	v->ng_gain = v->ng_gain*0.75f + tgain*0.25f;
 	if (!(++counter % 10))
 		ms_debug("%d: nglevel=%f, energy=%f, tgain=%f, ng_gain=%f",
 				          (v->peer!=NULL)?1:0, energy, v->energy, tgain, v->ng_gain);
@@ -261,7 +261,7 @@ static void volume_noise_gate_process(Volume *v , float energy, mblk_t *om){
 static int volume_set_db_gain(MSFilter *f, void *gain){
 	float *fgain=(float*)gain;
 	Volume *v=(Volume*)f->data;
-	v->gain = v->static_gain = pow(10,(*fgain)/10);
+	v->gain = v->static_gain = (float)pow(10,(*fgain)/10);
 	ms_message("MSVolume set gain to [%f db], [%f] linear",*fgain,v->gain);
 	return 0;
 }
@@ -398,8 +398,8 @@ static void update_energy(Volume *v, int16_t *signal, int numsamples, uint64_t c
 		if (lp > pk)
 			pk = lp;
 	}
-	en = (sqrt(acc / numsamples)+1) / max_e;
-	v->energy = (en * coef) + v->energy * (1.0 - coef);
+	en = (float)((sqrt(acc / numsamples)+1) / max_e);
+	v->energy = (en * coef) + v->energy * (1.0f - coef);
 	v->level_pk = (float)pk / max_e;
 	v->instant_energy = en;// currently non-averaged energy seems better (short artefacts)
 	ortp_extremum_record_max(&v->max,curtime,v->energy);
@@ -428,7 +428,7 @@ static void apply_gain(Volume *v, mblk_t *m, float tgain) {
 	}
 
 	gain=v->gain * v->ng_gain;
-	intgain = gain* 4096;
+	intgain = (int32_t)(gain* 4096);
 
 
 	//if (v->peer) ms_message("MSVolume:%p Applying gain %5f, v->gain=%5f, tgain=%5f, ng_gain=%5f",v,gain,v->gain,tgain,v->ng_gain);
@@ -441,7 +441,7 @@ static void apply_gain(Volume *v, mblk_t *m, float tgain) {
 			*sample = saturate(((*sample - v->dc_offset) * intgain) / 4096);
 		}
 		/* offset smoothing */
-		v->dc_offset = (v->dc_offset*7 + dc_offset*2/(m->b_wptr - m->b_rptr)) / 8;
+		v->dc_offset = (v->dc_offset*7 + dc_offset*2/(int)(m->b_wptr - m->b_rptr)) / 8;
 	}else if (gain!=1){
 		for (	sample=(int16_t*)m->b_rptr;
 					sample<(int16_t*)m->b_wptr;
@@ -512,7 +512,7 @@ static void volume_process(MSFilter *f){
 	}else{
 		/*light processing: no agc. Work in place in the input buffer*/
 		while((m=ms_queue_get(f->inputs[0]))!=NULL){
-			update_energy(v,(int16_t*)m->b_rptr, (m->b_wptr - m->b_rptr) / 2, f->ticker->time);
+			update_energy(v,(int16_t*)m->b_rptr, (int)((m->b_wptr - m->b_rptr) / 2), f->ticker->time);
 			target_gain = v->static_gain;
 
 			if (v->noise_gate_enabled)

src/audiofilters/speexec.c

@@ -47,7 +47,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #endif
 
 static const int framesize=64;
-static const int flow_control_interval_ms=5000;
+static const uint32_t flow_control_interval_ms=5000;
 
 
 typedef struct SpeexECState{
@@ -135,7 +135,7 @@ static void apply_config(SpeexECState *s){
 			ms_error("Could not decode base64 %s",s->state_str);
 			return;
 		}
-		blob=speex_echo_state_blob_new_from_memory(buffer,buflen);
+		blob=speex_echo_state_blob_new_from_memory(buffer,(int)buflen);
 		if (blob==NULL){
 			ms_error("Could not create blob from config string");
 			return;
@@ -263,7 +263,7 @@ static void speex_ec_process(MSFilter *f){
 		int avail_samples;
 
 		if (!s->echostarted) s->echostarted=TRUE;
-		if ((avail=ms_bufferizer_get_avail(&s->delayed_ref))<((s->nominal_ref_samples*2)+nbytes)){
+		if ((avail=(int)ms_bufferizer_get_avail(&s->delayed_ref))<((s->nominal_ref_samples*2)+nbytes)){
 			/*we don't have enough to read in a reference signal buffer, inject silence instead*/
 			avail=nbytes;
 			refm=allocb(nbytes,0);

src/audiofilters/tonedetector.c

@@ -33,7 +33,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 
 #define MAX_SCANS 10
 
-static const float energy_min_threshold=0.01;
+static const float energy_min_threshold=0.01f;
 
 typedef struct _GoertzelState{
 	uint64_t starttime;
@@ -64,7 +64,7 @@ static float goertzel_state_run(GoertzelState *gs,int16_t  *samples, int nsample
 	
 	freq_en= (q1*q1) + (q2*q2) - (q1*q2*gs->coef);
 	/*return a relative frequency energy compared over the total signal energy */
-	return freq_en/(total_energy*(float)nsamples*0.5);
+	return freq_en/(total_energy*(float)nsamples*0.5f);
 }
 
 static float compute_energy(int16_t *samples, int nsamples){

src/base/mscommon.c

@@ -376,7 +376,7 @@ void ms_usleep(uint64_t usec){
 #else
 	HANDLE sleepEvent = CreateEventEx(NULL, NULL, CREATE_EVENT_MANUAL_RESET, EVENT_ALL_ACCESS);
 	if (!sleepEvent) return;
-	WaitForSingleObjectEx(sleepEvent, usec / 1000, FALSE);
+	WaitForSingleObjectEx(sleepEvent, (DWORD)(usec / 1000), FALSE);
 #endif
 #else
 	struct timespec ts,rem;
@@ -445,7 +445,7 @@ int ms_concealer_inc_sample_time(MSConcealerContext* obj, uint64_t current_time,
 	}
 	obj->sample_time+=time_increment;
 	if (obj->plc_start_time!=-1 && got_packet){
-		plc_duration=current_time-obj->plc_start_time;
+		plc_duration=(int)(current_time-obj->plc_start_time);
 		obj->plc_start_time=-1;
 		if (plc_duration>obj->max_plc_time) plc_duration=obj->max_plc_time;
 	}

src/base/msqueue.c

@@ -87,7 +87,7 @@ int ms_bufferizer_read(MSBufferizer *obj, uint8_t *data, int datalen){
 		/* first store current meta information in the _q_stopper field the queue, just to reuse space*/
 		mblk_meta_copy(m, &obj->q._q_stopper);
 		while(sz<datalen){
-			cplen=MIN(m->b_wptr-m->b_rptr,datalen-sz);
+			cplen=MIN((int)(m->b_wptr-m->b_rptr),datalen-sz);
 			if (data) memcpy(data+sz,m->b_rptr,cplen);
 			sz+=cplen;
 			m->b_rptr+=cplen;

src/base/msticker.c

@@ -540,7 +540,7 @@ float ms_ticker_get_average_load(MSTicker *ticker){
 		once=TRUE;
 	}
 #endif
-	return ticker->av_load;
+	return (float)ticker->av_load;
 }
 
 

src/crypto/dtls_srtp.c

@@ -367,7 +367,7 @@ static int ms_dtls_srtp_rtp_sendData (void *ctx, const unsigned char *data, size
 	rtp_session_get_transports(session,&rtpt,NULL);
 
 	/* generate message from raw data */
-	msg = rtp_session_create_packet_raw((uint8_t *)data, (int)length);
+	msg = rtp_session_create_packet_raw((uint8_t *)data, length);
 
 	ret = meta_rtp_transport_modifier_inject_packet(rtpt, context->rtp_modifier, msg , 0);
 
@@ -402,7 +402,7 @@ static int ms_dtls_srtp_rtcp_sendData (void *ctx, const unsigned char *data, siz
 	rtp_session_get_transports(session,NULL,&rtcpt);
 
 	/* generate message from raw data */
-	msg = rtp_session_create_packet_raw((uint8_t *)data, (int)length);
+	msg = rtp_session_create_packet_raw((uint8_t *)data, length);
 
 	ret = meta_rtp_transport_modifier_inject_packet(rtcpt, context->rtcp_modifier, msg , 0);
 	freemsg(msg);

src/crypto/ms_srtp.c

@@ -107,7 +107,7 @@ static int _process_on_send(RtpSession* session,MSSrtpStreamContext *ctx, mblk_t
 	bool_t is_rtp=ctx->is_rtp;
 	rtp_header_t *rtp_header=is_rtp?(rtp_header_t*)m->b_rptr:NULL;
 	rtcp_common_header_t *rtcp_header=!is_rtp?(rtcp_common_header_t*)m->b_rptr:NULL;
-	slen=msgdsize(m);
+	slen=(int)msgdsize(m);
 
 	if (rtp_header && (slen>RTP_FIXED_HEADER_SIZE && rtp_header->version==2)) {
 		ms_mutex_lock(&ctx->mutex);
@@ -151,7 +151,7 @@ static int ms_srtp_process_on_send(RtpTransportModifier *t, mblk_t *m){
 }
 
 static int ms_srtp_process_dummy(RtpTransportModifier *t, mblk_t *m) {
-	return msgdsize(m);
+	return (int)msgdsize(m);
 }
 static int _process_on_receive(RtpSession* session,MSSrtpStreamContext *ctx, mblk_t *m, int err){
 	int slen;
@@ -179,7 +179,7 @@ static int _process_on_receive(RtpSession* session,MSSrtpStreamContext *ctx, mbl
 	}
 }
 static int ms_srtp_process_on_receive(RtpTransportModifier *t, mblk_t *m){
-	return _process_on_receive(t->session,(MSSrtpStreamContext*)t->data, m,msgdsize(m));
+	return _process_on_receive(t->session,(MSSrtpStreamContext*)t->data, m,(int)msgdsize(m));
 }
 
 /**** Session management functions ****/

src/otherfilters/msrtp.c

@@ -175,7 +175,7 @@ static int sender_unmute(MSFilter * f, void *arg)
 static int sender_set_relay_session_id(MSFilter *f, void*arg){
 	SenderData *d = (SenderData *) f->data;
 	const char *tmp=(const char *)arg;
-	d->relay_session_id_size=b64_decode(tmp, strlen(tmp), (void*)d->relay_session_id, (unsigned int)sizeof(d->relay_session_id));
+	d->relay_session_id_size=(int)b64_decode(tmp, strlen(tmp), (void*)d->relay_session_id, (unsigned int)sizeof(d->relay_session_id));
 	return 0;
 }
 
@@ -230,7 +230,7 @@ static uint32_t get_cur_timestamp(MSFilter * f, mblk_t *im){
 			d->tsoff = curts - packet_ts;
 		}else{
 			diffts=packet_ts-d->last_ts;
-			difftime_ts=((f->ticker->time-d->last_sent_time)*d->rate)/1000;
+			difftime_ts=(int)(((f->ticker->time-d->last_sent_time)*d->rate)/1000);
 			/* detect timestamp jump in the stream and adjust so that they become continuous on the network*/
 			if (abs(diffts-difftime_ts)>(d->rate/5)){
 				uint32_t tsoff=curts - packet_ts;

src/utils/_kiss_fft_guts.h

@@ -142,7 +142,7 @@ struct kiss_fft_state{
 #else
 #  define KISS_FFT_COS(phase) (kiss_fft_scalar) cos(phase)
 #  define KISS_FFT_SIN(phase) (kiss_fft_scalar) sin(phase)
-#  define HALF_OF(x) ((x)*.5)
+#  define HALF_OF(x) ((x)*.5f)
 #endif
 
 #define  kf_cexp(x,phase) \

src/utils/audiodiff.c

@@ -157,7 +157,7 @@ static int compute_cross_correlation(int16_t *s1, int n1, int16_t *s2_padded, fl
 	for (i=0; i<xcorr_nsamples; i++){
 		norm2 += s2_padded[step*(i+n1-1)]*s2_padded[step*(i+n1-1)];
 		tmp = scalar_product(s1, s2_padded + i*step, n1, step);
-		xcorr[i] = (double)tmp / sqrt((double)(norm1)*(double)norm2);
+		xcorr[i] = (float)((double)tmp / sqrt((double)(norm1)*(double)norm2));
 		tmp = tmp < 0 ? -tmp : tmp;
 		if (tmp > max){
 			max = tmp;
@@ -264,7 +264,7 @@ static int _ms_audio_diff_chunked(FileInfo *fi1, FileInfo *fi2, double *ret, int
 	num_chunks = i;
 	
 	ms_message("tot_energy is %li", (long int) tot_energy);
-	maxpos = cum_maxpos / tot_energy;
+	maxpos = (int)(cum_maxpos / tot_energy);