Add SSE2 support for Windows.

The previous defines controlling SSE2 use assumed GCC. To fix this:
- Import a chunk of defines to typedefs.h from Chrome's build_config.h, primarily to get WEBRTC_ARCH_X86_FAMILY.
- Add a check derived from WebP to define WEBRTC_USE_SSE2.
- Modify cpu_features.cc to work with MSVC. This code is borrowed from chrome/src/base/cpu.cc.
- Change AEC defines to use WEBRTC_USE_SSE2.
- Remove disable_sse2 check from aec.gyp. This is handled by WEBRTC_USE_SSE2.

(Also remove a bit of unused code from aec_core.h)
Review URL: http://webrtc-codereview.appspot.com/95008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@299 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/audio_processing/aec/main/source/aec.gyp

@@ -29,20 +29,14 @@
       'sources': [
         '../interface/echo_cancellation.h',
         'echo_cancellation.c',
+        'aec_core.h',
         'aec_core.c',
+        'aec_core_sse2.c',
         'aec_rdft.h',
         'aec_rdft.c',
-        'aec_core.h',
-        'resampler.c',
+        'aec_rdft_sse2.c',
         'resampler.h',
-      ],
-      'conditions': [
-        ['disable_sse2 == 0 and (target_arch == "ia32" or target_arch == "x64")', {
-          'sources': [
-            'aec_core_sse2.c',
-            'aec_rdft_sse2.c',
-          ],
-        }],
+        'resampler.c',
       ],
     },
   ],

src/modules/audio_processing/aec/main/source/aec_core.c

@@ -468,7 +468,7 @@ int WebRtcAec_InitAec(aec_t *aec, int sampFreq)
     WebRtcAec_FilterAdaptation = FilterAdaptation;
     WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
     if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
+#if defined(WEBRTC_USE_SSE2)
       WebRtcAec_InitAec_SSE2();
 #endif
     }
@@ -561,8 +561,8 @@ void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
 }
 
 static void ProcessBlock(aec_t *aec, const short *farend,
-                              const short *nearend, const short *nearendH,
-                              short *output, short *outputH)
+                         const short *nearend, const short *nearendH,
+                         short *output, short *outputH)
 {
     int i;
     float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
@@ -601,7 +601,6 @@ static void ProcessBlock(aec_t *aec, const short *farend,
         }
     }
 
-
     memcpy(fft, aec->xBuf, sizeof(float) * PART_LEN2);
     memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);
     // For H band

src/modules/audio_processing/aec/main/source/aec_core.h

@@ -16,8 +16,9 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
 
 #include <stdio.h>
-#include "typedefs.h"
+
 #include "signal_processing_library.h"
+#include "typedefs.h"
 
 //#define G167 // for running G167 tests
 //#define UNCONSTR // time-unconstrained filter
@@ -92,21 +93,13 @@ typedef struct {
     float dMinPow[PART_LEN1];
     float dInitMinPow[PART_LEN1];
     float *noisePow;
-#ifdef FFTW
-    float fftR[PART_LEN2];
-    fftw_complex fftC[PART_LEN2];
-    fftw_plan fftPlan, ifftPlan;
 
-    fftw_complex xfBuf[NR_PART * PART_LEN1];
-    fftw_complex wfBuf[NR_PART * PART_LEN1];
-    fftw_complex sde[PART_LEN1];
-#else
     float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
     float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
     complex_t sde[PART_LEN1]; // cross-psd of nearend and error
     complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
     complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
-#endif
+
     float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
     float hNs[PART_LEN1];
     float hNlFbMin, hNlFbLocalMin;
@@ -169,8 +162,6 @@ typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
 extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
 typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
 extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-#define IP_LEN PART_LEN // this must be at least ceil(2 + sqrt(PART_LEN))
-#define W_LEN PART_LEN
 typedef void (*WebRtcAec_FilterAdaptation_t)
   (aec_t *aec, float *fft, float ef[2][PART_LEN1]);
 extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;

src/modules/audio_processing/aec/main/source/aec_core_sse2.c

@@ -12,7 +12,9 @@
  * The core AEC algorithm, SSE2 version of speed-critical functions.
  */
 
-#if defined(__SSE2__)
+#include "typedefs.h"
+
+#if defined(WEBRTC_USE_SSE2)
 #include <emmintrin.h>
 #include <math.h>
 
@@ -210,14 +212,6 @@ static void FilterAdaptationSSE2(aec_t *aec, float *fft, float ef[2][PART_LEN1])
   }
 }
 
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
 static __m128 mm_pow_ps(__m128 a, __m128 b)
 {
   // a^b = exp2(b * log2(a))
@@ -432,4 +426,4 @@ void WebRtcAec_InitAec_SSE2(void) {
   WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
 }
 
-#endif   //__SSE2__
+#endif   // WEBRTC_USE_SSE2

src/modules/audio_processing/aec/main/source/aec_rdft.c

@@ -19,10 +19,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "aec_rdft.h"
+
 #include <math.h>
 
-#include "aec_rdft.h"
 #include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "typedefs.h"
 
 // constants shared by all paths (C, SSE2).
 float rdft_w[64];
@@ -571,7 +573,7 @@ void aec_rdft_init(void) {
   rftfsub_128 = rftfsub_128_C;
   rftbsub_128 = rftbsub_128_C;
   if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
+#if defined(WEBRTC_USE_SSE2)
     aec_rdft_init_sse2();
 #endif
   }

src/modules/audio_processing/aec/main/source/aec_rdft.h

@@ -8,6 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
+
 #ifdef _MSC_VER /* visual c++ */
 # define ALIGN16_BEG __declspec(align(16))
 # define ALIGN16_END
@@ -40,3 +43,5 @@ void aec_rdft_init(void);
 void aec_rdft_init_sse2(void);
 void aec_rdft_forward_128(float *a);
 void aec_rdft_inverse_128(float *a);
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_

src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c

@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#if defined(__SSE2__)
+#include "typedefs.h"
+
+#if defined(WEBRTC_USE_SSE2)
 #include <emmintrin.h>
 
 #include "aec_rdft.h"
@@ -261,4 +263,4 @@ void aec_rdft_init_sse2(void) {
   rftbsub_128 = rftbsub_128_SSE2;
 }
 
-#endif  // __SSE2__
+#endif  // WEBRTC_USE_SS2