From 9345e86551a0e59e77dfee6eed3e2fe2a833b269 Mon Sep 17 00:00:00 2001
From: Bjorn Volcker <bjornv@webrtc.org>
Date: Wed, 10 Jun 2015 21:43:36 +0200
Subject: [PATCH] audio_processing: Create now returns a pointer to the object

Affects
* NS
* AGC
* AEC

BUG=441
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1175903002.

Cr-Commit-Position: refs/heads/master@{#9411}
---
 .../modules/audio_processing/aec/aec_core.c   | 36 +++++++------------
 .../modules/audio_processing/aec/aec_core.h   |  2 +-
 .../audio_processing/aec/aec_resampler.c      | 10 ++----
 .../audio_processing/aec/aec_resampler.h      |  4 +--
 .../audio_processing/aec/echo_cancellation.c  | 33 +++++++----------
 .../aec/echo_cancellation_unittest.cc         | 12 +++----
 .../aec/include/echo_cancellation.h           | 15 ++------
 .../aec/system_delay_unittest.cc              |  4 ++-
 .../audio_processing/agc/legacy/analog_agc.c  | 28 +++++----------
 .../agc/legacy/gain_control.h                 |  9 ++---
 .../echo_cancellation_impl.cc                 |  9 +----
 .../audio_processing/gain_control_impl.cc     |  9 +----
 .../noise_suppression_impl.cc                 | 12 ++-----
 .../ns/include/noise_suppression.h            | 15 ++------
 .../ns/include/noise_suppression_x.h          | 15 ++------
 .../audio_processing/ns/noise_suppression.c   | 13 +++----
 .../audio_processing/ns/noise_suppression_x.c | 17 +++------
 17 files changed, 70 insertions(+), 173 deletions(-)

diff --git a/webrtc/modules/audio_processing/aec/aec_core.c b/webrtc/modules/audio_processing/aec/aec_core.c
index 32266f4e32..16e3389cef 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.c
+++ b/webrtc/modules/audio_processing/aec/aec_core.c
@@ -1367,26 +1367,23 @@ static void ProcessBlock(AecCore* aec) {
 #endif
 }
 
-int WebRtcAec_CreateAec(AecCore** aecInst) {
+AecCore* WebRtcAec_CreateAec() {
   int i;
   AecCore* aec = malloc(sizeof(AecCore));
-  *aecInst = aec;
-  if (aec == NULL) {
-    return -1;
+  if (!aec) {
+    return NULL;
   }
 
   aec->nearFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
   if (!aec->nearFrBuf) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
 
   aec->outFrBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(float));
   if (!aec->outFrBuf) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
 
   for (i = 0; i < NUM_HIGH_BANDS_MAX; ++i) {
@@ -1394,15 +1391,13 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
                                              sizeof(float));
     if (!aec->nearFrBufH[i]) {
       WebRtcAec_FreeAec(aec);
-      aec = NULL;
-      return -1;
+      return NULL;
     }
     aec->outFrBufH[i] = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
                                             sizeof(float));
     if (!aec->outFrBufH[i]) {
       WebRtcAec_FreeAec(aec);
-      aec = NULL;
-      return -1;
+      return NULL;
     }
   }
 
@@ -1411,15 +1406,13 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
       WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
   if (!aec->far_buf) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
   aec->far_buf_windowed =
       WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * 2 * PART_LEN1);
   if (!aec->far_buf_windowed) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
 #ifdef WEBRTC_AEC_DEBUG_DUMP
   aec->instance_index = webrtc_aec_instance_count;
@@ -1427,8 +1420,7 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
       WebRtc_CreateBuffer(kBufSizePartitions, sizeof(float) * PART_LEN);
   if (!aec->far_time_buf) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
   aec->farFile = aec->nearFile = aec->outFile = aec->outLinearFile = NULL;
   aec->debug_dump_count = 0;
@@ -1437,8 +1429,7 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
       WebRtc_CreateDelayEstimatorFarend(PART_LEN1, kHistorySizeBlocks);
   if (aec->delay_estimator_farend == NULL) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
   // We create the delay_estimator with the same amount of maximum lookahead as
   // the delay history size (kHistorySizeBlocks) for symmetry reasons.
@@ -1446,8 +1437,7 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
       aec->delay_estimator_farend, kHistorySizeBlocks);
   if (aec->delay_estimator == NULL) {
     WebRtcAec_FreeAec(aec);
-    aec = NULL;
-    return -1;
+    return NULL;
   }
 #ifdef WEBRTC_ANDROID
   // DA-AEC assumes the system is causal from the beginning and will self adjust
@@ -1485,7 +1475,7 @@ int WebRtcAec_CreateAec(AecCore** aecInst) {
 
   aec_rdft_init();
 
-  return 0;
+  return aec;
 }
 
 void WebRtcAec_FreeAec(AecCore* aec) {
diff --git a/webrtc/modules/audio_processing/aec/aec_core.h b/webrtc/modules/audio_processing/aec/aec_core.h
index aa5687a4f4..2fa26db859 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.h
+++ b/webrtc/modules/audio_processing/aec/aec_core.h
@@ -51,7 +51,7 @@ typedef struct Stats {
 
 typedef struct AecCore AecCore;
 
-int WebRtcAec_CreateAec(AecCore** aec);
+AecCore* WebRtcAec_CreateAec();  // Returns NULL on error.
 void WebRtcAec_FreeAec(AecCore* aec);
 int WebRtcAec_InitAec(AecCore* aec, int sampFreq);
 void WebRtcAec_InitAec_SSE2(void);
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.c b/webrtc/modules/audio_processing/aec/aec_resampler.c
index 8b07e16584..cef262edc1 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.c
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.c
@@ -40,14 +40,8 @@ static int EstimateSkew(const int* rawSkew,
                         int absLimit,
                         float* skewEst);
 
-int WebRtcAec_CreateResampler(void** resampInst) {
-  AecResampler* obj = malloc(sizeof(AecResampler));
-  *resampInst = obj;
-  if (obj == NULL) {
-    return -1;
-  }
-
-  return 0;
+void* WebRtcAec_CreateResampler() {
+  return malloc(sizeof(AecResampler));
 }
 
 int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz) {
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.h b/webrtc/modules/audio_processing/aec/aec_resampler.h
index debe069eee..a37499258f 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.h
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.h
@@ -20,8 +20,8 @@ enum {
   kResamplerBufferSize = FRAME_LEN * 4
 };
 
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcAec_CreateResampler(void** resampInst);
+// Unless otherwise specified, functions return 0 on success and -1 on error.
+void* WebRtcAec_CreateResampler();  // Returns NULL on error.
 int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
 void WebRtcAec_FreeResampler(void* resampInst);
 
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation.c b/webrtc/modules/audio_processing/aec/echo_cancellation.c
index 06d081d3ce..ec16aaf1d3 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation.c
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation.c
@@ -118,28 +118,22 @@ static void ProcessExtended(Aec* self,
                             int16_t reported_delay_ms,
                             int32_t skew);
 
-int32_t WebRtcAec_Create(void** aecInst) {
-  Aec* aecpc;
-  if (aecInst == NULL) {
-    return -1;
+void* WebRtcAec_Create() {
+  Aec* aecpc = malloc(sizeof(Aec));
+
+  if (!aecpc) {
+    return NULL;
   }
 
-  aecpc = malloc(sizeof(Aec));
-  *aecInst = aecpc;
-  if (aecpc == NULL) {
-    return -1;
-  }
-
-  if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
+  aecpc->aec = WebRtcAec_CreateAec();
+  if (!aecpc->aec) {
     WebRtcAec_Free(aecpc);
-    aecpc = NULL;
-    return -1;
+    return NULL;
   }
-
-  if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
+  aecpc->resampler = WebRtcAec_CreateResampler();
+  if (!aecpc->resampler) {
     WebRtcAec_Free(aecpc);
-    aecpc = NULL;
-    return -1;
+    return NULL;
   }
   // Create far-end pre-buffer. The buffer size has to be large enough for
   // largest possible drift compensation (kResamplerBufferSize) + "almost" an
@@ -148,8 +142,7 @@ int32_t WebRtcAec_Create(void** aecInst) {
       WebRtc_CreateBuffer(PART_LEN2 + kResamplerBufferSize, sizeof(float));
   if (!aecpc->far_pre_buf) {
     WebRtcAec_Free(aecpc);
-    aecpc = NULL;
-    return -1;
+    return NULL;
   }
 
   aecpc->initFlag = 0;
@@ -168,7 +161,7 @@ int32_t WebRtcAec_Create(void** aecInst) {
   }
 #endif
 
-  return 0;
+  return aecpc;
 }
 
 void WebRtcAec_Free(void* aecInst) {
diff --git a/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc b/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
index 6c325e0c51..315ac3e9f9 100644
--- a/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/echo_cancellation_unittest.cc
@@ -20,22 +20,20 @@ extern "C" {
 }
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 
 namespace webrtc {
 
 TEST(EchoCancellationTest, CreateAndFreeHasExpectedBehavior) {
-  EXPECT_EQ(-1, WebRtcAec_Create(NULL));
-  void* handle = NULL;
-  ASSERT_EQ(0, WebRtcAec_Create(&handle));
-  EXPECT_TRUE(handle != NULL);
+  void* handle = WebRtcAec_Create();
+  ASSERT_TRUE(handle);
   WebRtcAec_Free(nullptr);
   WebRtcAec_Free(handle);
 }
 
 TEST(EchoCancellationTest, ApplyAecCoreHandle) {
-  void* handle = NULL;
-  ASSERT_EQ(0, WebRtcAec_Create(&handle));
-  EXPECT_TRUE(handle != NULL);
+  void* handle = WebRtcAec_Create();
+  ASSERT_TRUE(handle);
   EXPECT_TRUE(WebRtcAec_aec_core(NULL) == NULL);
   AecCore* aec_core = WebRtcAec_aec_core(handle);
   EXPECT_TRUE(aec_core != NULL);
diff --git a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
index 6914ed1a9e..e49a0847e0 100644
--- a/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
+++ b/webrtc/modules/audio_processing/aec/include/echo_cancellation.h
@@ -64,19 +64,10 @@ extern "C" {
 
 /*
  * Allocates the memory needed by the AEC. The memory needs to be initialized
- * separately using the WebRtcAec_Init() function.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void**  aecInst              Pointer to the AEC instance to be created
- *                              and initialized
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * int32_t return               0: OK
- *                             -1: error
+ * separately using the WebRtcAec_Init() function. Returns a pointer to the
+ * object or NULL on error.
  */
-int32_t WebRtcAec_Create(void** aecInst);
+void* WebRtcAec_Create();
 
 /*
  * This function releases the memory allocated by WebRtcAec_Create().
diff --git a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
index da28752bcc..42800c50a3 100644
--- a/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
+++ b/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 }
@@ -67,7 +68,8 @@ SystemDelayTest::SystemDelayTest()
 }
 
 void SystemDelayTest::SetUp() {
-  ASSERT_EQ(0, WebRtcAec_Create(&handle_));
+  handle_ = WebRtcAec_Create();
+  ASSERT_TRUE(handle_);
   self_ = reinterpret_cast<Aec*>(handle_);
 }
 
diff --git a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
index bb56cfe95d..73adb5d3d2 100644
--- a/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
@@ -1313,31 +1313,19 @@ int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config) {
     return 0;
 }
 
-int WebRtcAgc_Create(void **agcInst)
-{
-  LegacyAgc* stt;
-    if (agcInst == NULL)
-    {
-        return -1;
-    }
-    stt = (LegacyAgc*)malloc(sizeof(LegacyAgc));
-
-    *agcInst = stt;
-    if (stt == NULL)
-    {
-        return -1;
-    }
+void* WebRtcAgc_Create() {
+  LegacyAgc* stt = malloc(sizeof(LegacyAgc));
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
-    stt->fpt = fopen("./agc_test_log.txt", "wt");
-    stt->agcLog = fopen("./agc_debug_log.txt", "wt");
-    stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
+  stt->fpt = fopen("./agc_test_log.txt", "wt");
+  stt->agcLog = fopen("./agc_debug_log.txt", "wt");
+  stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
 #endif
 
-    stt->initFlag = 0;
-    stt->lastError = 0;
+  stt->initFlag = 0;
+  stt->lastError = 0;
 
-    return 0;
+  return stt;
 }
 
 void WebRtcAgc_Free(void *state) {
diff --git a/webrtc/modules/audio_processing/agc/legacy/gain_control.h b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
index 3994f55e86..0ccba76c4b 100644
--- a/webrtc/modules/audio_processing/agc/legacy/gain_control.h
+++ b/webrtc/modules/audio_processing/agc/legacy/gain_control.h
@@ -200,13 +200,10 @@ int WebRtcAgc_set_config(void* agcInst, WebRtcAgcConfig config);
 int WebRtcAgc_get_config(void* agcInst, WebRtcAgcConfig* config);
 
 /*
- * This function creates an AGC instance, which will contain the state
- * information for one (duplex) channel.
- *
- * Return value             : AGC instance if successful
- *                          : 0 (i.e., a NULL pointer) if unsuccessful
+ * This function creates and returns an AGC instance, which will contain the
+ * state information for one (duplex) channel.
  */
-int WebRtcAgc_Create(void **agcInst);
+void* WebRtcAgc_Create();
 
 /*
  * This function frees the AGC instance created at the beginning.
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index e28f127839..fd3e1bf01c 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -337,14 +337,7 @@ void EchoCancellationImpl::SetExtraOptions(const Config& config) {
 }
 
 void* EchoCancellationImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAec_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
+  return WebRtcAec_Create();
 }
 
 void EchoCancellationImpl::DestroyHandle(void* handle) const {
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 5f301c1345..398cf5c5fd 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -301,14 +301,7 @@ int GainControlImpl::Initialize() {
 }
 
 void* GainControlImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAgc_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
+  return WebRtcAgc_Create();
 }
 
 void GainControlImpl::DestroyHandle(void* handle) const {
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index aa37e67fa8..1501911264 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -134,19 +134,11 @@ float NoiseSuppressionImpl::speech_probability() const {
 }
 
 void* NoiseSuppressionImpl::CreateHandle() const {
-  Handle* handle = NULL;
 #if defined(WEBRTC_NS_FLOAT)
-  if (WebRtcNs_Create(&handle) != apm_->kNoError)
+  return WebRtcNs_Create();
 #elif defined(WEBRTC_NS_FIXED)
-  if (WebRtcNsx_Create(&handle) != apm_->kNoError)
+  return WebRtcNsx_Create();
 #endif
-  {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
 }
 
 void NoiseSuppressionImpl::DestroyHandle(void* handle) const {
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression.h b/webrtc/modules/audio_processing/ns/include/noise_suppression.h
index 14e686a8e3..41bf9aca41 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression.h
+++ b/webrtc/modules/audio_processing/ns/include/noise_suppression.h
@@ -20,20 +20,9 @@ extern "C" {
 #endif
 
 /*
- * This function creates an instance to the noise suppression structure
- *
- * Input:
- *      - NS_inst       : Pointer to noise suppression instance that should be
- *                        created
- *
- * Output:
- *      - NS_inst       : Pointer to created noise suppression instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
+ * This function creates an instance of the floating point Noise Suppression.
  */
-int WebRtcNs_Create(NsHandle** NS_inst);
-
+NsHandle* WebRtcNs_Create();
 
 /*
  * This function frees the dynamic memory of a specified noise suppression
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h b/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
index 736cb23fa7..88fe4cd635 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
+++ b/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
@@ -20,20 +20,9 @@ extern "C" {
 #endif
 
 /*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- *      - nsxInst       : Pointer to noise reduction instance that should be
- *                       created
- *
- * Output:
- *      - nsxInst       : Pointer to created noise reduction instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
+ * This function creates an instance of the fixed point Noise Suppression.
  */
-int WebRtcNsx_Create(NsxHandle** nsxInst);
-
+NsxHandle* WebRtcNsx_Create();
 
 /*
  * This function frees the dynamic memory of a specified Noise Suppression
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression.c b/webrtc/modules/audio_processing/ns/noise_suppression.c
index 0efbebcc4e..cdecd62732 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression.c
@@ -17,15 +17,10 @@
 #include "webrtc/modules/audio_processing/ns/defines.h"
 #include "webrtc/modules/audio_processing/ns/ns_core.h"
 
-int WebRtcNs_Create(NsHandle** NS_inst) {
-  *NS_inst = (NsHandle*)malloc(sizeof(NoiseSuppressionC));
-  if (*NS_inst != NULL) {
-    (*(NoiseSuppressionC**)NS_inst)->initFlag = 0;
-    return 0;
-  } else {
-    return -1;
-  }
-
+NsHandle* WebRtcNs_Create() {
+  NoiseSuppressionC* self = malloc(sizeof(NoiseSuppressionC));
+  self->initFlag = 0;
+  return (NsHandle*)self;
 }
 
 void WebRtcNs_Free(NsHandle* NS_inst) {
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression_x.c b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
index a3b6d0f3bd..150fe608dd 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression_x.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
@@ -16,19 +16,12 @@
 #include "webrtc/modules/audio_processing/ns/nsx_core.h"
 #include "webrtc/modules/audio_processing/ns/nsx_defines.h"
 
-int WebRtcNsx_Create(NsxHandle** nsxInst) {
+NsxHandle* WebRtcNsx_Create() {
   NoiseSuppressionFixedC* self = malloc(sizeof(NoiseSuppressionFixedC));
-  *nsxInst = (NsxHandle*)self;
-
-  if (self != NULL) {
-    WebRtcSpl_Init();
-    self->real_fft = NULL;
-    self->initFlag = 0;
-    return 0;
-  } else {
-    return -1;
-  }
-
+  WebRtcSpl_Init();
+  self->real_fft = NULL;
+  self->initFlag = 0;
+  return (NsxHandle*)self;
 }
 
 void WebRtcNsx_Free(NsxHandle* nsxInst) {