Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/common_audio/audio_ring_buffer_unittest.cc

 /*
  *  Copyright (c) 2015 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 16 matching lines @@
   const size_t num_channels = input.num_channels();
   const size_t total_frames = input.num_frames();
   AudioRingBuffer buf(num_channels, buffer_frames);
   rtc::scoped_ptr<float* []> slice(new float* [num_channels]);
 
   size_t input_pos = 0;
   size_t output_pos = 0;
   while (input_pos + buf.WriteFramesAvailable() < total_frames) {
     // Write until the buffer is as full as possible.
     while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
-      buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
-                num_channels, num_write_chunk_frames);
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
       input_pos += num_write_chunk_frames;
     }
     // Read until the buffer is as empty as possible.
     while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
       EXPECT_LT(output_pos, total_frames);
-      buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
-               num_channels, num_read_chunk_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
       output_pos += num_read_chunk_frames;
     }
   }
 
   // Write and read the last bit.
   if (input_pos < total_frames) {
-    buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)),
-              num_channels, total_frames - input_pos);
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
   }
   if (buf.ReadFramesAvailable()) {
-    buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)),
-             num_channels, buf.ReadFramesAvailable());
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
   }
   EXPECT_EQ(0u, buf.ReadFramesAvailable());
 }
 
 TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
   const size_t kFrames = 5000;
   const size_t num_channels = ::testing::get<3>(GetParam());
 
   // Initialize the input data to an increasing sequence.
   ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
@@ skipping 33 matching lines @@
   buf.MoveReadPositionForward(3);
   ChannelBuffer<float> output(1, kNumChannels);
   buf.Read(output.channels(), kNumChannels, 1);
   EXPECT_EQ(4, output.channels()[0][0]);
   buf.MoveReadPositionBackward(3);
   buf.Read(output.channels(), kNumChannels, 1);
   EXPECT_EQ(2, output.channels()[0][0]);
 }
 
 }  // namespace webrtc