Update audio code to use size_t more correctly, webrtc/common_audio/ portion.

webrtc/common_audio/audio_converter.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
 #include "webrtc/common_audio/audio_converter.h"
 
 #include <cstring>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/system_wrappers/interface/scoped_vector.h"
 
 using rtc::checked_cast;
 
 namespace webrtc {
 
 class CopyConverter : public AudioConverter {
  public:
-  CopyConverter(int src_channels, int src_frames, int dst_channels,
-                int dst_frames)
+  CopyConverter(int src_channels, size_t src_frames, int dst_channels,
+                size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~CopyConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
     if (src != dst) {
       for (int i = 0; i < src_channels(); ++i)
         std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
     }
   }
 };
 
 class UpmixConverter : public AudioConverter {
  public:
-  UpmixConverter(int src_channels, int src_frames, int dst_channels,
-                 int dst_frames)
+  UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
+                 size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
   ~UpmixConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
-    for (int i = 0; i < dst_frames(); ++i) {
+    for (size_t i = 0; i < dst_frames(); ++i) {
       const float value = src[0][i];
       for (int j = 0; j < dst_channels(); ++j)
         dst[j][i] = value;
     }
   }
 };
 
 class DownmixConverter : public AudioConverter {
  public:
-  DownmixConverter(int src_channels, int src_frames, int dst_channels,
-                   int dst_frames)
+  DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
+                   size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
   }
   ~DownmixConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
     float* dst_mono = dst[0];
-    for (int i = 0; i < src_frames(); ++i) {
+    for (size_t i = 0; i < src_frames(); ++i) {
       float sum = 0;
       for (int j = 0; j < src_channels(); ++j)
         sum += src[j][i];
       dst_mono[i] = sum / src_channels();
     }
   }
 };
 
 class ResampleConverter : public AudioConverter {
  public:
-  ResampleConverter(int src_channels, int src_frames, int dst_channels,
-                    int dst_frames)
+  ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
+                    size_t dst_frames)
       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
     resamplers_.reserve(src_channels);
     for (int i = 0; i < src_channels; ++i)
       resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
   }
   ~ResampleConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
     CheckSizes(src_size, dst_capacity);
@@ skipping 34 matching lines @@
     converters_.back()->Convert(buffers_.back()->channels(),
                                 buffers_.back()->size(), dst, dst_capacity);
   }
 
  private:
   ScopedVector<AudioConverter> converters_;
   ScopedVector<ChannelBuffer<float>> buffers_;
 };
 
 rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
-                                                       int src_frames,
+                                                       size_t src_frames,
                                                        int dst_channels,
-                                                       int dst_frames) {
+                                                       size_t dst_frames) {
   rtc::scoped_ptr<AudioConverter> sp;
   if (src_channels > dst_channels) {
     if (src_frames != dst_frames) {
       ScopedVector<AudioConverter> converters;
       converters.push_back(new DownmixConverter(src_channels, src_frames,
                                                 dst_channels, src_frames));
       converters.push_back(new ResampleConverter(dst_channels, src_frames,
                                                  dst_channels, dst_frames));
       sp.reset(new CompositionConverter(converters.Pass()));
     } else {
@@ skipping 23 matching lines @@
   return sp.Pass();
 }
 
 // For CompositionConverter.
 AudioConverter::AudioConverter()
     : src_channels_(0),
       src_frames_(0),
       dst_channels_(0),
       dst_frames_(0) {}
 
-AudioConverter::AudioConverter(int src_channels, int src_frames,
-                               int dst_channels, int dst_frames)
+AudioConverter::AudioConverter(int src_channels, size_t src_frames,
+                               int dst_channels, size_t dst_frames)
     : src_channels_(src_channels),
       src_frames_(src_frames),
       dst_channels_(dst_channels),
       dst_frames_(dst_frames) {
   CHECK(dst_channels == src_channels || dst_channels == 1 || src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
-  CHECK_EQ(src_size, checked_cast<size_t>(src_channels() * src_frames()));
-  CHECK_GE(dst_capacity, checked_cast<size_t>(dst_channels() * dst_frames()));
+  CHECK_EQ(src_size, src_channels() * src_frames());
+  CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc