webrtc aec: use 10 ms frames at 8khz

src/android/androidsound.cpp

@@ -629,7 +629,7 @@ static void android_snd_write_cb(int event, void *user, void * p_info){
 			MSTimeSpec ts;
 			ms_get_cur_time(&ts);
 			ms_message("%03u.%03u: AudioTrack ask %d, given %d, available %d [%f ms]", (uint32_t) ts.tv_sec, (uint32_t) (ts.tv_nsec / 1000),
-				ask, info->size, avail, (avail / (2 * ad->nchannels)) / (ad->rate / 1000.0));
+				ask, info.size, avail, (avail / (2 * ad->nchannels)) / (ad->rate / 1000.0));
 		}
 #endif
 		if (info.size > 0){

src/audiofilters/devices.c

@@ -75,7 +75,7 @@ static SoundDeviceDescription devices[]={
 	{	"samsung",	"GT-N7100",		"",			DEVICE_HAS_BUILTIN_AEC,	0 },  /*Galaxy Note 2*/
 	{	"samsung",	"GT-N7105",		"",			DEVICE_HAS_BUILTIN_AEC,	0 },  /*Galaxy Note 2*/
 	{	"samsung",	"SGH-T889",		"",			DEVICE_HAS_BUILTIN_AEC,	0 },  /*Galaxy Note 2*/
-	{	"samsung",	"Nexus S",		"s5pc110",	0,	200 },	/*After 4.1, hardware AEC is enabled*/
+	{	"samsung",	"Nexus S",		"s5pc110",	DEVICE_HAS_BUILTIN_AEC_CRAPPY,	180 }, /*Nexus S gives calibration around 240ms, but in practice the internal buffer size shrinks after a couple of seconds.*/
 	{	"samsung",	"Galaxy Nexus", "",			0,	120 },
 	{	"samsung",	"GT-S5570I",	"",			0,	250},
 	{	"samsung",	"GT-P3100",		"",			DEVICE_HAS_BUILTIN_AEC, 0 }, /* Galaxy Tab*/
@@ -206,8 +206,8 @@ SoundDeviceDescription * sound_device_description_get(void){
 	}else d=&undefined;
 	
 	if (declares_builtin_aec){
-		if (exact_match && (d->flags & DEVICE_HAS_BUILTIN_AEC_LYING)){
-			ms_warning("This device declares a builtin AEC but according to internal tables it is known to be lying about this, so trusting tables.");
+		if (exact_match && (d->flags & DEVICE_HAS_BUILTIN_AEC_CRAPPY)){
+			ms_warning("This device declares a builtin AEC but according to internal tables it is known to be misfunctionning, so trusting tables.");
 		}else{
 			d->flags=DEVICE_HAS_BUILTIN_AEC;
 			d->delay=0;

src/audiofilters/devices.h

@@ -25,7 +25,7 @@
 #include <mediastreamer2/mscommon.h>
 
 #define DEVICE_HAS_BUILTIN_AEC 			(1)
-#define DEVICE_HAS_BUILTIN_AEC_LYING	(1<<1) /*set when the device is claiming to have AEC but we should not trust it */
+#define DEVICE_HAS_BUILTIN_AEC_CRAPPY	(1<<1) /*set when the device is claiming to have AEC but we should not trust it */
 
 struct SoundDeviceDescription{
 	const char *manufacturer;

src/audiofilters/webrtc_aec.c

@@ -129,7 +129,7 @@ static mblk_t *audio_flow_controller_process(AudioFlowController *ctl, mblk_t *m
 #endif
 
 static const float smooth_factor = 0.05;
-static const int framesize = 160;
+static const int framesize = 80;
 static const int flow_control_interval_ms = 5000;
 
 
@@ -212,22 +212,27 @@ static void webrtc_aec_preprocess(MSFilter *f)
 
 	s->echostarted = FALSE;
 	delay_samples = s->delay_ms * s->samplerate / 1000;
+	s->framesize=(framesize*s->samplerate)/8000;
 	ms_message("Initializing WebRTC echo canceler with framesize=%i, delay_ms=%i, delay_samples=%i", s->framesize, s->delay_ms, delay_samples);
 
 	if (WebRtcAecm_Create(&s->aecmInst) < 0) {
 		s->bypass_mode = TRUE;
+		ms_error("WebRtcAecm_Create(): error, entering bypass mode");
 		return;
 	}
 	if (WebRtcAecm_Init(s->aecmInst, s->samplerate) < 0) {
 		if (WebRtcAecm_get_error_code(s->aecmInst) == AECM_BAD_PARAMETER_ERROR) {
-			ms_error("WebRTC echo canceller does not support %d samplerate", s->samplerate);
+			ms_error("WebRtcAecm_Init(): WebRTC echo canceller does not support %d samplerate", s->samplerate);
 		}
 		s->bypass_mode = TRUE;
+		ms_error("Entering bypass mode");
 		return;
 	}
 	config.cngMode = TRUE;
 	config.echoMode = 3;
-	WebRtcAecm_set_config(s->aecmInst, config);
+	if (WebRtcAecm_set_config(s->aecmInst, config)!=0){
+		ms_error("WebRtcAecm_set_config(): failed.");
+	}
 
 	/* fill with zeroes for the time of the delay*/
 	m = allocb(delay_samples * 2, 0);
@@ -325,8 +330,10 @@ static void webrtc_aec_process(MSFilter *f)
 		if (s->echofile)
 			fwrite(echo, nbytes, 1, s->echofile);
 #endif
-		WebRtcAecm_BufferFarend(s->aecmInst, (const WebRtc_Word16 *) ref, s->framesize);
-		WebRtcAecm_Process(s->aecmInst, (const WebRtc_Word16 *) echo, NULL, (WebRtc_Word16 *) oecho->b_wptr, s->framesize, 0);
+		if (WebRtcAecm_BufferFarend(s->aecmInst, (const WebRtc_Word16 *) ref, s->framesize)!=0)
+			ms_error("WebRtcAecm_BufferFarend() failed.");
+		if (WebRtcAecm_Process(s->aecmInst, (const WebRtc_Word16 *) echo, NULL, (WebRtc_Word16 *) oecho->b_wptr, s->framesize, 0)!=0)
+			ms_error("WebRtcAecm_Process() failed.");
 #ifdef EC_DUMP
 		if (s->cleanfile)
 			fwrite(oecho->b_wptr, nbytes, 1, s->cleanfile);
@@ -364,13 +371,26 @@ static void webrtc_aec_postprocess(MSFilter *f)
 static int webrtc_aec_set_sr(MSFilter *f, void *arg)
 {
 	WebRTCAECState *s = (WebRTCAECState *) f->data;
-	s->samplerate = *(int *) arg;
+	int sr=*(int *) arg;
+	
+	if (sr!=8000 && sr!=16000){
+		ms_message("Webrtc aec does not support sampling rate %i",sr);
+		return -1;
+	}
+	s->samplerate = sr;
+	return 0;
+}
+
+static int webrtc_aec_get_sr(MSFilter *f, void *arg)
+{
+	WebRTCAECState *s = (WebRTCAECState *) f->data;
+	*(int *) arg=s->samplerate;
 	return 0;
 }
 
 static int webrtc_aec_set_framesize(MSFilter *f, void *arg)
 {
-	/* Do nothing because the WebRTC echo canceller only accept specific values: 80 and 160. We use 160. */
+	/* Do nothing because the WebRTC echo canceller only accept specific values: 80 and 160. We use 80 at 8khz, and 160 at 16khz */
 	return 0;
 }
 
@@ -416,6 +436,7 @@ static int webrtc_aec_get_state(MSFilter *f, void *arg)
 
 static MSFilterMethod webrtc_aec_methods[] = {
 	{	MS_FILTER_SET_SAMPLE_RATE		,	webrtc_aec_set_sr 		},
+	{	MS_FILTER_GET_SAMPLE_RATE		,	webrtc_aec_get_sr 		},
 	{	MS_ECHO_CANCELLER_SET_TAIL_LENGTH	,	webrtc_aec_set_tail_length	},
 	{	MS_ECHO_CANCELLER_SET_DELAY		,	webrtc_aec_set_delay		},
 	{	MS_ECHO_CANCELLER_SET_FRAMESIZE		,	webrtc_aec_set_framesize	},

src/voip/audiostream.c

@@ -319,11 +319,11 @@ int audio_stream_start_full(AudioStream *stream, RtpProfile *profile, const char
 		picker_context.picker=&audio_stream_payload_picker;
 		ms_filter_call_method(stream->ms.decoder,MS_FILTER_SET_RTP_PAYLOAD_PICKER, &picker_context);
 	}
-	if((stream->features & AUDIO_STREAM_FEATURE_VOL_SND) != 0)
+	if ((stream->features & AUDIO_STREAM_FEATURE_VOL_SND) != 0)
 		stream->volsend=ms_filter_new(MS_VOLUME_ID);
 	else
 		stream->volsend=NULL;
-	if((stream->features & AUDIO_STREAM_FEATURE_VOL_RCV) != 0)
+	if ((stream->features & AUDIO_STREAM_FEATURE_VOL_RCV) != 0)
 		stream->volrecv=ms_filter_new(MS_VOLUME_ID);
 	else
 		stream->volrecv=NULL;