Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/opus/opus_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/opus/interface/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
 
 using ::std::string;
 
 namespace webrtc {
 
 static const int kOpusBlockDurationMs = 20;
 static const int kOpusSamplingKhz = 48;
 
 class OpusSpeedTest : public AudioCodecSpeedTest {
  protected:
   OpusSpeedTest();
   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);
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 };
 
 OpusSpeedTest::OpusSpeedTest()
     : AudioCodecSpeedTest(kOpusBlockDurationMs,
                           kOpusSamplingKhz,
                           kOpusSamplingKhz),
       opus_encoder_(NULL),
@@ skipping 12 matching lines @@
 }
 
 void OpusSpeedTest::TearDown() {
   AudioCodecSpeedTest::TearDown();
   /* Free memory. */
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
 
 float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
-                                  int max_bytes, int* encoded_bytes) {
+                                  size_t max_bytes, size_t* encoded_bytes) {
   clock_t clocks = clock();
   int value = WebRtcOpus_Encode(opus_encoder_, in_data,
                                 input_length_sample_, max_bytes,
                                 bit_stream);
   clocks = clock() - clocks;
   EXPECT_GT(value, 0);
-  *encoded_bytes = value;
+  *encoded_bytes = static_cast<size_t>(value);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
 float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
-                                  int encoded_bytes, int16_t* out_data) {
+                                  size_t encoded_bytes, int16_t* out_data) {
   int value;
   int16_t audio_type;
   clock_t clocks = clock();
   value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
                             &audio_type);
   clocks = clock() - clocks;
   EXPECT_EQ(output_length_sample_, value);
   return 1000.0 * clocks / CLOCKS_PER_SEC;
 }
 
@@ skipping 28 matching lines @@
                             string("audio_coding/speech_mono_32_48kHz"),
                             string("pcm"), true),
      ::std::tr1::make_tuple(2, 64000,
                             string("audio_coding/music_stereo_48kHz"),
                             string("pcm"), true)};
 
 INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc