Remove typedefs.h from webrtc/ root (part 1)

webrtc/common_audio/resampler/sinc_resampler.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 142 matching lines @@
       input_buffer_size_(request_frames_ + kKernelSize),
       // Create input buffers with a 16-byte alignment for SSE optimizations.
       kernel_storage_(static_cast<float*>(
           AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
       kernel_pre_sinc_storage_(static_cast<float*>(
           AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
       kernel_window_storage_(static_cast<float*>(
           AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
       input_buffer_(static_cast<float*>(
           AlignedMalloc(sizeof(float) * input_buffer_size_, 16))),
-#if defined(WEBRTC_CPU_DETECTION)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
       convolve_proc_(nullptr),
 #endif
       r1_(input_buffer_.get()),
       r2_(input_buffer_.get() + kKernelSize / 2) {
-#if defined(WEBRTC_CPU_DETECTION)
+#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
   InitializeCPUSpecificFeatures();
   RTC_DCHECK(convolve_proc_);
 #endif
   RTC_DCHECK_GT(request_frames_, 0);
   Flush();
   RTC_DCHECK_GT(block_size_, kKernelSize);
 
   memset(kernel_storage_.get(), 0,
          sizeof(*kernel_storage_.get()) * kKernelStorageSize);
   memset(kernel_pre_sinc_storage_.get(), 0,
@@ skipping 186 matching lines @@
     sum1 += *input_ptr * *k1++;
     sum2 += *input_ptr++ * *k2++;
   }
 
   // Linearly interpolate the two "convolutions".
   return static_cast<float>((1.0 - kernel_interpolation_factor) * sum1 +
       kernel_interpolation_factor * sum2);
 }
 
 }  // namespace webrtc