Noise suppression: Change signature to work on floats instead of ints

Internally, it already worked on floats. This patch just changes the
signature of a bunch of functions so that floats can be passed
directly from the new and improved AudioBuffer without converting the
data to int and back again first.

(The reference data to the ApmTest.Process test had to be modified
slightly; this is because the noise suppressor comes immediately after
the echo canceller, which also works on floats. If I truncate to
integers between the two steps, ApmTest.Process doesn't complain, but
of course that's exactly the sort of thing the float conversion is
supposed to let us avoid...)

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6385 4adac7df-926f-26a2-2b94-8c16560cd09d

5 files changed, 36 insertions, 70 deletions

diff --git a/modules/audio_processing/noise_suppression_impl.cc b/modules/audio_processing/noise_suppression_impl.cc
index c3eb7b01..eea0a04a 100644
--- a/modules/audio_processing/noise_suppression_impl.cc
+++ b/modules/audio_processing/noise_suppression_impl.cc
@@ -68,10 +68,10 @@ int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
 #if defined(WEBRTC_NS_FLOAT)
     err = WebRtcNs_Process(static_cast<Handle*>(handle(i)),
-                           audio->low_pass_split_data(i),
-                           audio->high_pass_split_data(i),
-                           audio->low_pass_split_data(i),
-                           audio->high_pass_split_data(i));
+                           audio->low_pass_split_data_f(i),
+                           audio->high_pass_split_data_f(i),
+                           audio->low_pass_split_data_f(i),
+                           audio->high_pass_split_data_f(i));
 #elif defined(WEBRTC_NS_FIXED)
     err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)),
                             audio->low_pass_split_data(i),
diff --git a/modules/audio_processing/ns/include/noise_suppression.h b/modules/audio_processing/ns/include/noise_suppression.h
index 32b18038..3cf889e2 100644
--- a/modules/audio_processing/ns/include/noise_suppression.h
+++ b/modules/audio_processing/ns/include/noise_suppression.h
@@ -99,10 +99,10 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode);
  *                        -1 - Error
  */
 int WebRtcNs_Process(NsHandle* NS_inst,
-                     short* spframe,
-                     short* spframe_H,
-                     short* outframe,
-                     short* outframe_H);
+                     float* spframe,
+                     float* spframe_H,
+                     float* outframe,
+                     float* outframe_H);
 
 /* Returns the internally used prior speech probability of the current frame.
  * There is a frequency bin based one as well, with which this should not be
diff --git a/modules/audio_processing/ns/noise_suppression.c b/modules/audio_processing/ns/noise_suppression.c
index 848467f0..075ab88c 100644
--- a/modules/audio_processing/ns/noise_suppression.c
+++ b/modules/audio_processing/ns/noise_suppression.c
@@ -43,8 +43,8 @@ int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
 }
 
 
-int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H,
-                     short* outframe, short* outframe_H) {
+int WebRtcNs_Process(NsHandle* NS_inst, float* spframe, float* spframe_H,
+                     float* outframe, float* outframe_H) {
   return WebRtcNs_ProcessCore(
       (NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
 }
diff --git a/modules/audio_processing/ns/ns_core.c b/modules/audio_processing/ns/ns_core.c
index 124a66d8..ec267ae0 100644
--- a/modules/audio_processing/ns/ns_core.c
+++ b/modules/audio_processing/ns/ns_core.c
@@ -715,10 +715,10 @@ void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snr
 }
 
 int WebRtcNs_ProcessCore(NSinst_t* inst,
-                         short* speechFrame,
-                         short* speechFrameHB,
-                         short* outFrame,
-                         short* outFrameHB) {
+                         float* speechFrame,
+                         float* speechFrameHB,
+                         float* outFrame,
+                         float* outFrameHB) {
   // main routine for noise reduction
 
   int     flagHB = 0;
@@ -731,8 +731,8 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
   float   snrPrior, currentEstimateStsa;
   float   tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
   float   gammaNoiseTmp, gammaNoiseOld;
-  float   noiseUpdateTmp, fTmp, dTmp;
-  float   fin[BLOCKL_MAX], fout[BLOCKL_MAX];
+  float   noiseUpdateTmp, fTmp;
+  float   fout[BLOCKL_MAX];
   float   winData[ANAL_BLOCKL_MAX];
   float   magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
   float   theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
@@ -775,26 +775,17 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
   updateParsFlag = inst->modelUpdatePars[0];
   //
 
-  //for LB do all processing
-  // convert to float
-  for (i = 0; i < inst->blockLen10ms; i++) {
-    fin[i] = (float)speechFrame[i];
-  }
   // update analysis buffer for L band
   memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
          sizeof(float) * (inst->anaLen - inst->blockLen10ms));
-  memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
+  memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, speechFrame,
          sizeof(float) * inst->blockLen10ms);
 
   if (flagHB == 1) {
-    // convert to float
-    for (i = 0; i < inst->blockLen10ms; i++) {
-      fin[i] = (float)speechFrameHB[i];
-    }
     // update analysis buffer for H band
     memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
            sizeof(float) * (inst->anaLen - inst->blockLen10ms));
-    memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
+    memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, speechFrameHB,
            sizeof(float) * inst->blockLen10ms);
   }
 
@@ -833,30 +824,16 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
           inst->outBuf[i] = fout[i + inst->blockLen10ms];
         }
       }
-      // convert to short
-      for (i = 0; i < inst->blockLen10ms; i++) {
-        dTmp = fout[i];
-        if (dTmp < WEBRTC_SPL_WORD16_MIN) {
-          dTmp = WEBRTC_SPL_WORD16_MIN;
-        } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
-          dTmp = WEBRTC_SPL_WORD16_MAX;
-        }
-        outFrame[i] = (short)dTmp;
-      }
+      for (i = 0; i < inst->blockLen10ms; ++i)
+        outFrame[i] = WEBRTC_SPL_SAT(
+            WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
 
       // for time-domain gain of HB
-      if (flagHB == 1) {
-        for (i = 0; i < inst->blockLen10ms; i++) {
-          dTmp = inst->dataBufHB[i];
-          if (dTmp < WEBRTC_SPL_WORD16_MIN) {
-            dTmp = WEBRTC_SPL_WORD16_MIN;
-          } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
-            dTmp = WEBRTC_SPL_WORD16_MAX;
-          }
-          outFrameHB[i] = (short)dTmp;
-        }
-      }  // end of H band gain computation
-      //
+      if (flagHB == 1)
+        for (i = 0; i < inst->blockLen10ms; ++i)
+          outFrameHB[i] = WEBRTC_SPL_SAT(
+              WEBRTC_SPL_WORD16_MAX, inst->dataBufHB[i], WEBRTC_SPL_WORD16_MIN);
+
       return 0;
     }
 
@@ -1239,16 +1216,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
     inst->outLen -= inst->blockLen10ms;
   }
 
-  // convert to short
-  for (i = 0; i < inst->blockLen10ms; i++) {
-    dTmp = fout[i];
-    if (dTmp < WEBRTC_SPL_WORD16_MIN) {
-      dTmp = WEBRTC_SPL_WORD16_MIN;
-    } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
-      dTmp = WEBRTC_SPL_WORD16_MAX;
-    }
-    outFrame[i] = (short)dTmp;
-  }
+  for (i = 0; i < inst->blockLen10ms; ++i)
+    outFrame[i] = WEBRTC_SPL_SAT(
+        WEBRTC_SPL_WORD16_MAX, fout[i], WEBRTC_SPL_WORD16_MIN);
 
   // for time-domain gain of HB
   if (flagHB == 1) {
@@ -1289,13 +1259,9 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
     }
     //apply gain
     for (i = 0; i < inst->blockLen10ms; i++) {
-      dTmp = gainTimeDomainHB * inst->dataBufHB[i];
-      if (dTmp < WEBRTC_SPL_WORD16_MIN) {
-        dTmp = WEBRTC_SPL_WORD16_MIN;
-      } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
-        dTmp = WEBRTC_SPL_WORD16_MAX;
-      }
-      outFrameHB[i] = (short)dTmp;
+      float o = gainTimeDomainHB * inst->dataBufHB[i];
+      outFrameHB[i] = WEBRTC_SPL_SAT(
+          WEBRTC_SPL_WORD16_MAX, o, WEBRTC_SPL_WORD16_MIN);
     }
   }  // end of H band gain computation
   //
diff --git a/modules/audio_processing/ns/ns_core.h b/modules/audio_processing/ns/ns_core.h
index 50daa137..785239eb 100644
--- a/modules/audio_processing/ns/ns_core.h
+++ b/modules/audio_processing/ns/ns_core.h
@@ -167,10 +167,10 @@ int WebRtcNs_set_policy_core(NSinst_t* inst, int mode);
 
 
 int WebRtcNs_ProcessCore(NSinst_t* inst,
-                         short* inFrameLow,
-                         short* inFrameHigh,
-                         short* outFrameLow,
-                         short* outFrameHigh);
+                         float* inFrameLow,
+                         float* inFrameHigh,
+                         float* outFrameLow,
+                         float* outFrameHigh);
 
 
 #ifdef __cplusplus