More work on the AEC (including changes in PJSUA), embed the AEC in sound_port, reduce DirectSound buffer from 32 to 16, and fixed ARM compilation for MSVC WinCE target.

git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@648 74dad513-b988-da41-8d7b-12977e46ad98

7 files changed, 159 insertions, 73 deletions

diff --git a/pjmedia/include/pjmedia/aec.h b/pjmedia/include/pjmedia/aec.h
index a2354ef3..84ed7e0f 100644
--- a/pjmedia/include/pjmedia/aec.h
+++ b/pjmedia/include/pjmedia/aec.h
@@ -51,14 +51,14 @@ typedef struct pjmedia_aec pjmedia_aec;
  * @param pool		    Pool to allocate memory.
  * @param clock_rate	    Media clock rate/sampling rate.
  * @param samples_per_frame Number of samples per frame.
- * @param tail_size	    Tail length, in number of samples.
+ * @param tail_ms	    Tail length, miliseconds.
  *
  * @return		    PJ_SUCCESS on success.
  */
 PJ_DECL(pj_status_t) pjmedia_aec_create( pj_pool_t *pool,
 					 unsigned clock_rate,
 					 unsigned samples_per_frame,
-					 unsigned tail_size,
+					 unsigned tail_ms,
 					 unsigned options,
 					 pjmedia_aec **p_aec );
 
diff --git a/pjmedia/include/pjmedia/aec_port.h b/pjmedia/include/pjmedia/aec_port.h
index f18b1378..2968c915 100644
--- a/pjmedia/include/pjmedia/aec_port.h
+++ b/pjmedia/include/pjmedia/aec_port.h
@@ -43,14 +43,14 @@ PJ_BEGIN_DECL
  *
  * @param pool		Pool to allocate memory.
  * @param dn_port	Downstream port.
- * @param tail_length	Tail length in samples.
+ * @param tail_ms	Tail length in miliseconds.
  * @param p_port	Pointer to receive the port instance.
  *
  * @return		PJ_SUCCESS on success.
  */
 PJ_DECL(pj_status_t) pjmedia_aec_port_create( pj_pool_t *pool,
 					      pjmedia_port *dn_port,
-					      unsigned tail_length,
+					      unsigned tail_ms,
 					      pjmedia_port **p_port );
 
 
diff --git a/pjmedia/include/pjmedia/sound_port.h b/pjmedia/include/pjmedia/sound_port.h
index 15339bcc..c144275d 100644
--- a/pjmedia/include/pjmedia/sound_port.h
+++ b/pjmedia/include/pjmedia/sound_port.h
@@ -183,6 +183,40 @@ PJ_DECL(pjmedia_snd_stream*) pjmedia_snd_port_get_snd_stream(
 
 
 /**
+ * Enable accoustic echo cancellation (AEC) to the specified sound. 
+ * The AEC can only be enabled for sound streams with full-duplex direction.
+ *
+ * And note, you should only change the AEC settings when the sound port
+ * is not connected to any downstream ports.
+ *
+ * @param snd_port	    The sound device port.
+ * @param pool		    Pool to re-create the AEC if necessary.
+ * @param tail_ms	    Maximum echo tail length to be supported, in
+ *			    miliseconds. If zero is specified, the AEC would
+ *			    be disabled.
+ *
+ * @return		    PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjmedia_snd_port_set_aec(pjmedia_snd_port *snd_port,
+					      pj_pool_t *pool,
+					      unsigned tail_ms);
+
+
+/**
+ * Get current AEC tail length, in miliseconds. The tail length will be zero
+ * if AEC is not enabled.
+ *
+ * @param snd_port	    The sound device port.
+ * @param p_length	    Pointer to receive the tail length.
+ *
+ * @return		    PJ_SUCCESS on success.
+ */
+PJ_DECL(pj_status_t) pjmedia_snd_port_get_aec_tail(pjmedia_snd_port *snd_port,
+						   unsigned *p_length);
+
+
+
+/**
  * Connect a port to the sound device port. If the sound device port has a
  * sound recorder device, then this will start periodic function call to
  * the port's put_frame() function. If the sound device has a sound player
diff --git a/pjmedia/src/pjmedia/aec_port.c b/pjmedia/src/pjmedia/aec_port.c
index aa56b175..70d3a217 100644
--- a/pjmedia/src/pjmedia/aec_port.c
+++ b/pjmedia/src/pjmedia/aec_port.c
@@ -45,7 +45,7 @@ static pj_status_t aec_on_destroy(pjmedia_port *this_port);
 
 PJ_DEF(pj_status_t) pjmedia_aec_port_create( pj_pool_t *pool,
 					     pjmedia_port *dn_port,
-					     unsigned tail_length,
+					     unsigned tail_ms,
 					     pjmedia_port **p_port )
 {
     const pj_str_t AEC = { "AEC", 3 };
@@ -53,7 +53,7 @@ PJ_DEF(pj_status_t) pjmedia_aec_port_create( pj_pool_t *pool,
     pj_status_t status;
 
     PJ_ASSERT_RETURN(pool && dn_port && p_port, PJ_EINVAL);
-    PJ_ASSERT_RETURN(dn_port->info.bits_per_sample==16 && tail_length, 
+    PJ_ASSERT_RETURN(dn_port->info.bits_per_sample==16 && tail_ms, 
 		     PJ_EINVAL);
 
     /* Create the port and the AEC itself */
@@ -67,7 +67,7 @@ PJ_DEF(pj_status_t) pjmedia_aec_port_create( pj_pool_t *pool,
 
     status = pjmedia_aec_create(pool, dn_port->info.clock_rate, 
 				dn_port->info.samples_per_frame,
-				tail_length, 0, &aec->aec);
+				tail_ms, 0, &aec->aec);
     if (status != PJ_SUCCESS)
 	return status;
 
diff --git a/pjmedia/src/pjmedia/aec_speex.c b/pjmedia/src/pjmedia/aec_speex.c
index 76a8dd44..95cc15b4 100644
--- a/pjmedia/src/pjmedia/aec_speex.c
+++ b/pjmedia/src/pjmedia/aec_speex.c
@@ -26,10 +26,11 @@
 #include <pj/os.h>
 #include <pj/pool.h>
 #include <speex/speex_echo.h>
+#include <speex/speex_preprocess.h>
 
 
 #define THIS_FILE   "aec_speex.c"
-#define BUF_COUNT   16
+#define BUF_COUNT   8
 
 
 struct frame
@@ -39,10 +40,13 @@ struct frame
 
 struct pjmedia_aec
 {
-    SpeexEchoState  *state;
+    SpeexEchoState	 *state;
+    SpeexPreprocessState *preprocess;
+
     unsigned	     samples_per_frame;
     unsigned	     options;
     pj_int16_t	    *tmp_frame;
+    spx_int32_t	    *residue;
 
     pj_lock_t	    *lock;		/* To protect buffers, if required  */
 
@@ -59,7 +63,7 @@ struct pjmedia_aec
 PJ_DEF(pj_status_t) pjmedia_aec_create( pj_pool_t *pool,
 					unsigned clock_rate,
 					unsigned samples_per_frame,
-					unsigned tail_size,
+					unsigned tail_ms,
 					unsigned options,
 					pjmedia_aec **p_aec )
 {
@@ -68,6 +72,8 @@ PJ_DEF(pj_status_t) pjmedia_aec_create( pj_pool_t *pool,
     unsigned i;
     pj_status_t status;
 
+    *p_aec = NULL;
+
     aec = pj_pool_zalloc(pool, sizeof(pjmedia_aec));
     PJ_ASSERT_RETURN(aec != NULL, PJ_ENOMEM);
 
@@ -78,22 +84,35 @@ PJ_DEF(pj_status_t) pjmedia_aec_create( pj_pool_t *pool,
     aec->samples_per_frame = samples_per_frame;
     aec->options = options;
 
-    aec->state = speex_echo_state_init(samples_per_frame,tail_size);
+    aec->state = speex_echo_state_init(samples_per_frame,
+					clock_rate * tail_ms / 1000);
     if (aec->state == NULL) {
 	pj_lock_destroy(aec->lock);
 	return PJ_ENOMEM;
     }
 
+    aec->preprocess = speex_preprocess_state_init(samples_per_frame, 
+						  clock_rate);
+    if (aec->preprocess == NULL) {
+	speex_echo_state_destroy(aec->state);
+	pj_lock_destroy(aec->lock);
+	return PJ_ENOMEM;
+    }
+
     /* Set sampling rate */
     sampling_rate = clock_rate;
     speex_echo_ctl(aec->state, SPEEX_ECHO_SET_SAMPLING_RATE, 
 		   &sampling_rate);
 
     /* Create temporary frame for echo cancellation */
-    aec->tmp_frame = pj_pool_zalloc(pool, sizeof(pj_int16_t) *
-					    samples_per_frame);
+    aec->tmp_frame = pj_pool_zalloc(pool, 2 * samples_per_frame);
     PJ_ASSERT_RETURN(aec->tmp_frame != NULL, PJ_ENOMEM);
 
+    /* Create temporary frame to receive residue */
+    aec->residue = pj_pool_zalloc(pool, sizeof(spx_int32_t) * 
+					    samples_per_frame);
+    PJ_ASSERT_RETURN(aec->residue != NULL, PJ_ENOMEM);
+
     /* Create internal playback buffers */
     for (i=0; i<BUF_COUNT; ++i) {
 	aec->frames[i].buf = pj_pool_zalloc(pool, samples_per_frame * 2);
@@ -108,7 +127,7 @@ PJ_DEF(pj_status_t) pjmedia_aec_create( pj_pool_t *pool,
 			 "samples per frame=%d, tail length=%d ms", 
 			 clock_rate,
 			 samples_per_frame,
-			 tail_size * 1000 / clock_rate));
+			 tail_ms));
     return PJ_SUCCESS;
 
 }
@@ -121,11 +140,19 @@ PJ_DEF(pj_status_t) pjmedia_aec_destroy(pjmedia_aec *aec )
 {
     PJ_ASSERT_RETURN(aec && aec->state, PJ_EINVAL);
 
+    if (aec->lock)
+	pj_lock_acquire(aec->lock);
+
     if (aec->state) {
 	speex_echo_state_destroy(aec->state);
 	aec->state = NULL;
     }
 
+    if (aec->preprocess) {
+	speex_preprocess_state_destroy(aec->preprocess);
+	aec->preprocess = NULL;
+    }
+
     if (aec->lock) {
 	pj_lock_destroy(aec->lock);
 	aec->lock = NULL;
@@ -224,8 +251,6 @@ PJ_DEF(pj_status_t) pjmedia_aec_cancel_echo( pjmedia_aec *aec,
 					     unsigned options,
 					     void *reserved )
 {
-    unsigned level0, level1;
-
     /* Sanity checks */
     PJ_ASSERT_RETURN(aec && rec_frm && play_frm && options==0 &&
 		     reserved==NULL, PJ_EINVAL);
@@ -233,20 +258,13 @@ PJ_DEF(pj_status_t) pjmedia_aec_cancel_echo( pjmedia_aec *aec,
     /* Cancel echo, put output in temporary buffer */
     speex_echo_cancel(aec->state, (const spx_int16_t*)rec_frm, 
 		      (const spx_int16_t*)play_frm, 
-		      (spx_int16_t*)aec->tmp_frame, NULL);
+		      (spx_int16_t*)aec->tmp_frame, 
+		      aec->residue);
 
-#if 0
-    level0 = pjmedia_calc_avg_signal(rec_frm, aec->samples_per_frame);
-    level1 = pjmedia_calc_avg_signal(aec->tmp_frame, aec->samples_per_frame);
 
-    if (level1 < level0) {
-	PJ_LOG(5,(THIS_FILE, "Input signal reduced from %d to %d",
-		  level0, level1));
-    }
-#else
-    PJ_UNUSED_ARG(level0);
-    PJ_UNUSED_ARG(level1);
-#endif
+    /* Preprocess output */
+    speex_preprocess(aec->preprocess, (spx_int16_t*)aec->tmp_frame, 
+		     aec->residue);
 
     /* Copy temporary buffer back to original rec_frm */
     pjmedia_copy_samples(rec_frm, aec->tmp_frame, aec->samples_per_frame);
diff --git a/pjmedia/src/pjmedia/dsound.c b/pjmedia/src/pjmedia/dsound.c
index 762378f0..0f9c49a1 100644
--- a/pjmedia/src/pjmedia/dsound.c
+++ b/pjmedia/src/pjmedia/dsound.c
@@ -452,7 +452,6 @@ static int dsound_dev_thread(void *arg)
     HANDLE events[2];
     unsigned eventCount;
     unsigned bytes_per_frame;
-    int excess_rec = 0;
     pj_status_t status;
 
 
@@ -502,7 +501,6 @@ static int dsound_dev_thread(void *arg)
 	if (signalled_dir == PJMEDIA_DIR_PLAYBACK) {
 	    
 	    struct dsound_stream *dsound_strm;
-	    int i;
 
 	    /*
 	     * DirectSound has requested us to feed some frames to
@@ -512,31 +510,26 @@ static int dsound_dev_thread(void *arg)
 	    dsound_strm = &strm->play_strm;
 	    status = PJ_SUCCESS;
 
-	    for (i=0; i <= excess_rec; ++i) {
-		/* Get frame from application. */
-		status = (*strm->play_cb)(strm->user_data, 
-					  dsound_strm->timestamp.u32.lo,
-					  strm->buffer,
-					  bytes_per_frame);
-		if (status != PJ_SUCCESS)
-		    break;
-
-		/* Write to DirectSound buffer. */
-		AppWriteDataToBuffer( dsound_strm->ds.play.lpDsBuffer, 
-				      dsound_strm->dwBytePos,
-				      (LPBYTE)strm->buffer, 
+	    /* Get frame from application. */
+	    status = (*strm->play_cb)(strm->user_data, 
+				      dsound_strm->timestamp.u32.lo,
+				      strm->buffer,
 				      bytes_per_frame);
-
-		/* Increment position. */
-		dsound_strm->dwBytePos += bytes_per_frame;
-		if (dsound_strm->dwBytePos >= dsound_strm->dwDsBufferSize)
-		    dsound_strm->dwBytePos -= dsound_strm->dwDsBufferSize;
-		dsound_strm->timestamp.u64 += strm->samples_per_frame;
-	    }
-
 	    if (status != PJ_SUCCESS)
 		break;
 
+	    /* Write to DirectSound buffer. */
+	    AppWriteDataToBuffer( dsound_strm->ds.play.lpDsBuffer, 
+				  dsound_strm->dwBytePos,
+				  (LPBYTE)strm->buffer, 
+				  bytes_per_frame);
+
+	    /* Increment position. */
+	    dsound_strm->dwBytePos += bytes_per_frame;
+	    if (dsound_strm->dwBytePos >= dsound_strm->dwDsBufferSize)
+		dsound_strm->dwBytePos -= dsound_strm->dwDsBufferSize;
+	    dsound_strm->timestamp.u64 += strm->samples_per_frame;
+
 	} else {
 	    /*
 	     * DirectSound has indicated that it has some frames ready
@@ -544,7 +537,6 @@ static int dsound_dev_thread(void *arg)
 	     * prevent overflows.
 	     */
 	    struct dsound_stream *dsound_strm;
-	    int captured = 0;
 	    BOOL rc;
 
 	    dsound_strm = &strm->rec_strm;
@@ -558,9 +550,7 @@ static int dsound_dev_thread(void *arg)
 		
 		if (!rc) {
 		    pj_bzero(strm->buffer, bytes_per_frame);
-		} else {
-		    captured++;
-		}
+		} 
 
 		/* Call callback */
 		status = (*strm->rec_cb)(strm->user_data, 
@@ -582,8 +572,6 @@ static int dsound_dev_thread(void *arg)
 		/* Fetch while we have more than 1 frame */
 	    } while (dsound_captured_size(dsound_strm) > bytes_per_frame);
 
-	    excess_rec = captured-1;
-	    if (excess_rec < 0) excess_rec = 0;
 	}
     }
 
diff --git a/pjmedia/src/pjmedia/sound_port.c b/pjmedia/src/pjmedia/sound_port.c
index ee3e61c9..37338704 100644
--- a/pjmedia/src/pjmedia/sound_port.c
+++ b/pjmedia/src/pjmedia/sound_port.c
@@ -34,7 +34,7 @@
 #endif
 
 //#define SIMULATE_LOST_PCT   20
-#define AEC_TAIL    500	    /* in ms */
+#define AEC_TAIL    128	    /* default AEC length in ms */
 
 #define THIS_FILE	    "sound_port.c"
 
@@ -57,6 +57,7 @@ struct pjmedia_snd_port
     unsigned		 options;
 
     pjmedia_aec		*aec;
+    unsigned		 aec_tail_len;
     pjmedia_plc		*plc;
 
     unsigned		 clock_rate;
@@ -135,10 +136,6 @@ no_frame:
     }
 
 
-    if (snd_port->aec) {
-	pjmedia_aec_playback(snd_port->aec, output);
-    }
-
     return PJ_SUCCESS;
 }
 
@@ -156,11 +153,6 @@ static pj_status_t rec_cb(/* in */   void *user_data,
     pjmedia_port *port;
     pjmedia_frame frame;
 
-    /* Cancel echo */
-    if (snd_port->aec) {
-	pjmedia_aec_capture(snd_port->aec, input, 0);
-    }
-
     /* We're risking accessing the port without holding any mutex.
      * It's possible that port is disconnected then destroyed while
      * we're trying to access it.
@@ -171,6 +163,11 @@ static pj_status_t rec_cb(/* in */   void *user_data,
     if (port == NULL)
 	return PJ_SUCCESS;
 
+    /* Cancel echo */
+    if (snd_port->aec) {
+	pjmedia_aec_capture(snd_port->aec, input, 0);
+    }
+
     frame.buf = (void*)input;
     frame.size = size;
     frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
@@ -246,18 +243,19 @@ static pj_status_t start_sound_device( pj_pool_t *pool,
 				    snd_port->samples_per_frame * 
 					snd_port->channel_count,
 				    0, &snd_port->plc);
-	if (status != PJ_SUCCESS)
+	if (status != PJ_SUCCESS) {
+	    PJ_LOG(4,(THIS_FILE, "Unable to create PLC"));
 	    snd_port->plc = NULL;
+	}
     }
 
     /* Create AEC only when direction is full duplex */
     if (snd_port->dir == PJMEDIA_DIR_CAPTURE_PLAYBACK) {
-	status = pjmedia_aec_create(pool, snd_port->clock_rate, 
-				    snd_port->samples_per_frame, 
-				    snd_port->clock_rate * AEC_TAIL / 1000,
-				    0, &snd_port->aec);
-	if (status != PJ_SUCCESS)
+	status = pjmedia_snd_port_set_aec(snd_port, pool, AEC_TAIL);
+	if (status != PJ_SUCCESS) {
+	    PJ_LOG(4,(THIS_FILE, "Unable to create AEC"));
 	    snd_port->aec = NULL;
+	}
     }
 
     /* Start sound stream. */
@@ -432,6 +430,54 @@ PJ_DEF(pjmedia_snd_stream*) pjmedia_snd_port_get_snd_stream(
 
 
 /*
+ * Enable AEC
+ */
+PJ_DEF(pj_status_t) pjmedia_snd_port_set_aec( pjmedia_snd_port *snd_port,
+					      pj_pool_t *pool,
+					      unsigned tail_ms)
+{
+    pj_status_t status;
+
+    /* Sound must be opened in full-duplex mode */
+    PJ_ASSERT_RETURN(snd_port && 
+		     snd_port->dir == PJMEDIA_DIR_CAPTURE_PLAYBACK,
+		     PJ_EINVALIDOP);
+
+    /* Destroy AEC */
+    if (snd_port->aec) {
+	pjmedia_aec_destroy(snd_port->aec);
+	snd_port->aec = NULL;
+    }
+
+    snd_port->aec_tail_len = tail_ms;
+
+    if (tail_ms != 0) {
+	status = pjmedia_aec_create(pool, snd_port->clock_rate, 
+				    snd_port->samples_per_frame, 
+				    snd_port->clock_rate * tail_ms / 1000,
+				    0, &snd_port->aec);
+	if (status != PJ_SUCCESS)
+	    snd_port->aec = NULL;
+    } else {
+	status = PJ_SUCCESS;
+    }
+
+    return status;
+}
+
+
+/* Get AEC tail length */
+PJ_DEF(pj_status_t) pjmedia_snd_port_get_aec_tail( pjmedia_snd_port *snd_port,
+						   unsigned *p_length)
+{
+    PJ_ASSERT_RETURN(snd_port && p_length, PJ_EINVAL);
+    *p_length =  snd_port->aec ? snd_port->aec_tail_len : 0;
+    return PJ_SUCCESS;
+}
+
+
+
+/*
  * Connect a port.
  */
 PJ_DEF(pj_status_t) pjmedia_snd_port_connect( pjmedia_snd_port *snd_port,