This CL introduces namespaces in the aec c++ files

(the ones that were recently moved from c)

There are many files changed but most changes just
consist of adding namespaces.

In aec_common.h an C++-specific #ifdef needed to be added as
that file is both included from C and C++. I could see no
way around that but please let me know if there is a better
way around that.

BUG=webrtc:5201

Review URL: https://codereview.webrtc.org/1766663002

Cr-Commit-Position: refs/heads/master@{#11883}

webrtc/modules/audio_processing/aec/aec_common.h

@@ -21,6 +21,10 @@
 #define ALIGN16_END __attribute__((aligned(16)))
 #endif
 
+#ifdef __cplusplus
+namespace webrtc {
+#endif
+
 extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_sqrtHanning[65];
 extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_weightCurve[65];
 extern ALIGN16_BEG const float ALIGN16_END WebRtcAec_overDriveCurve[65];
@@ -28,4 +32,8 @@ extern const float WebRtcAec_kExtendedSmoothingCoefficients[2][2];
 extern const float WebRtcAec_kNormalSmoothingCoefficients[2][2];
 extern const float WebRtcAec_kMinFarendPSD;
 
+#ifdef __cplusplus
+}  // namespace webrtc
+#endif
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_

webrtc/modules/audio_processing/aec/aec_core.cc

@@ -40,6 +40,8 @@ extern "C" {
 #include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
+
 // Buffer size (samples)
 static const size_t kBufSizePartitions = 250;  // 1 second of audio in 16 kHz.
 
@@ -1901,3 +1903,4 @@ void WebRtcAec_SetSystemDelay(AecCore* self, int delay) {
   assert(delay >= 0);
   self->system_delay = delay;
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/aec_core.h

@@ -19,6 +19,8 @@
 
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
+
 #define FRAME_LEN 80
 #define PART_LEN 64               // Length of partition
 #define PART_LEN1 (PART_LEN + 1)  // Unique fft coefficients
@@ -132,4 +134,6 @@ int WebRtcAec_system_delay(AecCore* self);
 // care.
 void WebRtcAec_SetSystemDelay(AecCore* self, int delay);
 
+}  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_H_

webrtc/modules/audio_processing/aec/aec_core_internal.h

@@ -19,6 +19,8 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
+
 // Number of partitions for the extended filter mode. The first one is an enum
 // to be used in array declarations, as it represents the maximum filter length.
 enum { kExtendedNumPartitions = 32 };
@@ -227,4 +229,6 @@ extern WebRtcAecStoreAsComplex WebRtcAec_StoreAsComplex;
 typedef void (*WebRtcAecWindowData)(float* x_windowed, const float* x);
 extern WebRtcAecWindowData WebRtcAec_WindowData;
 
+}  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_CORE_INTERNAL_H_

webrtc/modules/audio_processing/aec/aec_core_mips.cc

@@ -24,6 +24,8 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
 }
 
+namespace webrtc {
+
 extern const float WebRtcAec_weightCurve[65];
 extern const float WebRtcAec_overDriveCurve[65];
 
@@ -781,3 +783,4 @@ void WebRtcAec_InitAec_mips(void) {
   WebRtcAec_ComfortNoise = WebRtcAec_ComfortNoise_mips;
   WebRtcAec_OverdriveAndSuppress = WebRtcAec_OverdriveAndSuppress_mips;
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/aec_core_neon.cc

@@ -27,6 +27,8 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
 }
 
+namespace webrtc {
+
 enum { kShiftExponentIntoTopMantissa = 8 };
 enum { kFloatExponentShift = 23 };
 
@@ -728,3 +730,4 @@ void WebRtcAec_InitAec_neon(void) {
   WebRtcAec_PartitionDelay = PartitionDelayNEON;
   WebRtcAec_WindowData = WindowDataNEON;
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/aec_core_sse2.cc

@@ -25,6 +25,8 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/aec_rdft.h"
 }
 
+namespace webrtc {
+
 __inline static float MulRe(float aRe, float aIm, float bRe, float bIm) {
   return aRe * bRe - aIm * bIm;
 }
@@ -742,3 +744,4 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_PartitionDelay = PartitionDelaySSE2;
   WebRtcAec_WindowData = WindowDataSSE2;
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/aec_resampler.cc

@@ -21,6 +21,8 @@
 
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
+namespace webrtc {
+
 enum { kEstimateLengthFrames = 400 };
 
 typedef struct {
@@ -202,3 +204,4 @@ int EstimateSkew(const int* rawSkew,
   *skewEst = skew;
   return 0;
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/aec_resampler.h

@@ -13,6 +13,8 @@
 
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
+namespace webrtc {
+
 enum { kResamplingDelay = 1 };
 enum { kResamplerBufferSize = FRAME_LEN * 4 };
 
@@ -32,4 +34,6 @@ void WebRtcAec_ResampleLinear(void* resampInst,
                               float* outspeech,
                               size_t* size_out);
 
+}  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_

webrtc/modules/audio_processing/aec/echo_cancellation.cc

@@ -29,6 +29,8 @@ extern "C" {
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
+
 // Measured delays [ms]
 // Device                Chrome  GTP
 // MacBook Air           10
@@ -881,3 +883,4 @@ static void EstBufDelayExtended(Aec* self) {
     self->knownDelay = WEBRTC_SPL_MAX((int)self->filtDelay - 256, 0);
   }
 }
+}  // namespace webrtc

webrtc/modules/audio_processing/aec/echo_cancellation.h

@@ -15,6 +15,8 @@
 
 #include "webrtc/typedefs.h"
 
+namespace webrtc {
+
 // Errors
 #define AEC_UNSPECIFIED_ERROR 12000
 #define AEC_UNSUPPORTED_FUNCTION_ERROR 12001
@@ -234,4 +236,6 @@ int WebRtcAec_GetDelayMetrics(void* handle,
 //
 struct AecCore* WebRtcAec_aec_core(void* handle);
 
+}  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_

webrtc/modules/audio_processing/aec/echo_cancellation_internal.h

@@ -16,6 +16,8 @@ extern "C" {
 }
 #include "webrtc/modules/audio_processing/aec/aec_core.h"
 
+namespace webrtc {
+
 typedef struct {
   int delayCtr;
   int sampFreq;
@@ -64,4 +66,6 @@ typedef struct {
   AecCore* aec;
 } Aec;
 
+}  // namespace webrtc
+
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_INTERNAL_H_

webrtc/modules/audio_processing/aec/system_delay_unittest.cc

@@ -13,7 +13,7 @@
 #include "webrtc/modules/audio_processing/aec/echo_cancellation_internal.h"
 #include "webrtc/modules/audio_processing/aec/echo_cancellation.h"
 #include "webrtc/typedefs.h"
-
+namespace webrtc {
 namespace {
 
 class SystemDelayTest : public ::testing::Test {
@@ -597,3 +597,4 @@ TEST_F(SystemDelayTest, CorrectImpactWhenTogglingDeviceBufferValues) {
 }
 
 }  // namespace
+}  // namespace webrtc

webrtc/modules/audio_processing/include/audio_processing.h

@@ -25,10 +25,10 @@
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/typedefs.h"
 
-struct AecCore;
-
 namespace webrtc {
 
+struct AecCore;
+
 class AudioFrame;
 
 template<typename T>