Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/opus/opus_unittest.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.
  */
 #include <string>
@@ skipping 88 matching lines @@
 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
                            const int16_t* input_audio,
                            const int input_samples,
                            WebRtcOpusDecInst* decoder,
                            int16_t* output_audio,
                            int16_t* audio_type) {
   encoded_bytes_ = WebRtcOpus_Encode(encoder,
                                     input_audio,
                                     input_samples, kMaxBytes,
                                     bitstream_);
+  EXPECT_GE(encoded_bytes_, 0);
   return WebRtcOpus_Decode(decoder, bitstream_,
                            encoded_bytes_, output_audio,
                            audio_type);
 }
 
 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
 // they should not. This test is signal dependent.
 void OpusTest::TestDtxEffect(bool dtx) {
   PrepareSpeechData(channels_, 20, 2000);
 
@@ skipping 413 matching lines @@
   EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
                                         channels_,
                                         application_));
   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
 
   // 10 ms. We use only first 10 ms of a 20 ms block.
   encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
                                      speech_data_.GetNextBlock(),
                                      kOpus10msFrameSamples, kMaxBytes,
                                      bitstream_);
+  EXPECT_GE(encoded_bytes_, 0);
   EXPECT_EQ(kOpus10msFrameSamples,
             WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
                                    encoded_bytes_));
 
   // 20 ms
   encoded_bytes_ = WebRtcOpus_Encode(opus_encoder_,
                                      speech_data_.GetNextBlock(),
                                      kOpus20msFrameSamples, kMaxBytes,
                                      bitstream_);
+  EXPECT_GE(encoded_bytes_, 0);
   EXPECT_EQ(kOpus20msFrameSamples,
             WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
                                    encoded_bytes_));
 
   // Free memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
 
 TEST_P(OpusTest, DISABLED_ON_IOS(OpusDecodeRepacketized)) {
@@ skipping 43 matching lines @@
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
 
 INSTANTIATE_TEST_CASE_P(VariousMode,
                         OpusTest,
                         Combine(Values(1, 2), Values(0, 1)));
 
 
 }  // namespace webrtc