From 12cccddc63cf7c94eb39e3c96e46e421f8eb9c00 Mon Sep 17 00:00:00 2001
From: "bjornv@webrtc.org"
 <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Thu, 19 Jan 2012 08:56:38 +0000
Subject: [PATCH] NS-SWB: Actived SWB processing at once, i.e., no startup
 phase.

Performance verified on a few 32 kHz files.
BUG=
TEST=audioproc, audioproc_unittest

Updated output_data_float.pb
Changes in SWB tests (3, 6, 9 and 12) as

Running test 3 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 1363
Expected: test->max_output_average()
Which is: 1386

Running test 6 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 2070
Expected: test->max_output_average()
Which is: 2109

Running test 9 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 1314
Expected: test->max_output_average()
Which is: 1336

Running test 12 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
  Actual: 2049
Expected: test->max_output_average()
Which is: 2085
Review URL: https://webrtc-codereview.appspot.com/344013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1465 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 src/modules/audio_processing/ns/ns_core.c     |  48 +++++++++---------
 .../audio_processing/output_data_float.pb     | Bin 1404 -> 1404 bytes
 2 files changed, 23 insertions(+), 25 deletions(-)

diff --git a/src/modules/audio_processing/ns/ns_core.c b/src/modules/audio_processing/ns/ns_core.c
index e80f699c5b..e5aa4ff364 100644
--- a/src/modules/audio_processing/ns/ns_core.c
+++ b/src/modules/audio_processing/ns/ns_core.c
@@ -1254,31 +1254,29 @@ int WebRtcNs_ProcessCore(NSinst_t* inst,
     for (i = 0; i < inst->magnLen; i++) {
       inst->speechProbHB[i] = probSpeechFinal[i];
     }
-    if (inst->blockInd > END_STARTUP_LONG) {
-      // average speech prob from low band
-      // avg over second half (i.e., 4->8kHz) of freq. spectrum
-      avgProbSpeechHB = 0.0;
-      for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) {
-        avgProbSpeechHB += inst->speechProbHB[i];
-      }
-      avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
-      // average filter gain from low band
-      // average over second half (i.e., 4->8kHz) of freq. spectrum
-      avgFilterGainHB = 0.0;
-      for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++) {
-        avgFilterGainHB += inst->smooth[i];
-      }
-      avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
-      avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
-      // gain based on speech prob:
-      gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
-      //combine gain with low band gain
-      gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
-      if (avgProbSpeechHB >= (float)0.5) {
-        gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
-      }
-      gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
-    } // end of converged
+    // average speech prob from low band
+    // avg over second half (i.e., 4->8kHz) of freq. spectrum
+    avgProbSpeechHB = 0.0;
+    for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) {
+      avgProbSpeechHB += inst->speechProbHB[i];
+    }
+    avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
+    // average filter gain from low band
+    // average over second half (i.e., 4->8kHz) of freq. spectrum
+    avgFilterGainHB = 0.0;
+    for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++) {
+      avgFilterGainHB += inst->smooth[i];
+    }
+    avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
+    avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
+    // gain based on speech prob:
+    gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
+    //combine gain with low band gain
+    gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
+    if (avgProbSpeechHB >= (float)0.5) {
+      gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
+    }
+    gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
     //make sure gain is within flooring range
     // flooring bottom
     if (gainTimeDomainHB < inst->denoiseBound) {
diff --git a/test/data/audio_processing/output_data_float.pb b/test/data/audio_processing/output_data_float.pb
index ccd7509267064cc5238c272cdfd2a12801e56505..6a48193d4a2e110d28d8c9ce0ee124ef9ef0ff6c 100644
GIT binary patch
delta 43
zcmeyv^@nT1C-%!+4$oOUeosEb?6BFA(T<UE+U5!-Uq;47n~yPvF)}tzmSv3x0E+t$
A%>V!Z

delta 43
zcmeyv^@nT1C-zrd4$oOUeosEb?6BFA(T<UE@8$|7Uq;3qn~yPvF)}WlEXx`X0Fu}c
AMF0Q*