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

webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.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/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
 #include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 using ::std::string;
 
 namespace webrtc {
 
 static const int kIsacBlockDurationMs = 30;
 static const int kIsacInputSamplingKhz = 16;
 static const int kIsacOutputSamplingKhz = 16;
 
 class IsacSpeedTest : public AudioCodecSpeedTest {
  protected:
   IsacSpeedTest();
   void SetUp() override;
   void TearDown() override;
   virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                             int max_bytes, int* encoded_bytes);
-  virtual float DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
+                             size_t max_bytes, size_t* encoded_bytes);
+  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
                              int16_t* out_data);
   ISACFIX_MainStruct *ISACFIX_main_inst_;
 };
 
 IsacSpeedTest::IsacSpeedTest()
     : AudioCodecSpeedTest(kIsacBlockDurationMs,
                           kIsacInputSamplingKhz,
                           kIsacOutputSamplingKhz),
       ISACFIX_main_inst_(NULL) {
 }
 
 void IsacSpeedTest::SetUp() {
   AudioCodecSpeedTest::SetUp();
 
   // Check whether the allocated buffer for the bit stream is large enough.
-  EXPECT_GE(max_bytes_, STREAM_MAXW16_60MS);
+  EXPECT_GE(max_bytes_, static_cast<size_t>(STREAM_MAXW16_60MS));
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_));
   EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1));
   EXPECT_EQ(0, WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_));
   // Set bitrate and block length.
   EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_,
                                      block_duration_ms_));
 }
 
 void IsacSpeedTest::TearDown() {
   AudioCodecSpeedTest::TearDown();
   // Free memory.
   EXPECT_EQ(0, WebRtcIsacfix_Free(ISACFIX_main_inst_));
 }
 
 float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                                  int max_bytes, int* encoded_bytes) {
+                                  size_t max_bytes, size_t* encoded_bytes) {
   // ISAC takes 10 ms everycall
   const int subblocks = block_duration_ms_ / 10;
   const int subblock_length = 10 * input_sampling_khz_;
   int value = 0;
 
   clock_t clocks = clock();
   size_t pointer = 0;
   for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
     value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
                                  bit_stream);
     if (idx == subblocks - 1)
       EXPECT_GT(value, 0);
     else
       EXPECT_EQ(0, value);
   }
   clocks = clock() - clocks;
-  *encoded_bytes = value;
+  *encoded_bytes = static_cast<size_t>(value);
   assert(*encoded_bytes <= max_bytes);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
 float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
-                                  int encoded_bytes,
+                                  size_t encoded_bytes,
                                   int16_t* out_data) {
   int value;
   int16_t audio_type;
   clock_t clocks = clock();
   value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
                                out_data, &audio_type);
   clocks = clock() - clocks;
   EXPECT_EQ(output_length_sample_, value);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
 TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) {
   size_t kDurationSec = 400;  // Test audio length in second.
   EncodeDecode(kDurationSec);
 }
 
 const coding_param param_set[] =
     {::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
                             string("pcm"), true)};
 
 INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc