Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.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.
  */
 
@@ skipping 118 matching lines @@
       << " ms when using VAD/CNG.";
 
   // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
   // following split sizes:
   // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
   // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
   int blocks_in_first_vad_call =
       (frames_to_encode > 3 ? 3 : frames_to_encode);
   if (frames_to_encode == 4)
     blocks_in_first_vad_call = 2;
+  CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
   const int blocks_in_second_vad_call =
       frames_to_encode - blocks_in_first_vad_call;
-  CHECK_GE(blocks_in_second_vad_call, 0);
 
   // Check if all of the buffer is passive speech. Start with checking the first
   // block.
   Vad::Activity activity = vad_->VoiceActivity(
       &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call,
       SampleRateHz());
   if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) {
     // Only check the second block if the first was passive.
     activity = vad_->VoiceActivity(
         &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call],
@@ skipping 58 matching lines @@
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
     int frames_to_encode,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
   for (int i = 0; i < frames_to_encode; ++i) {
     info = speech_encoder_->Encode(
         rtp_timestamps_.front(), &speech_buffer_[i * samples_per_10ms_frame],
         samples_per_10ms_frame, max_encoded_bytes, encoded);
-    if (i == frames_to_encode - 1) {
+    if (i + 1 == frames_to_encode) {
       CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
     } else {
       CHECK_EQ(info.encoded_bytes, 0u) << "Encoder delivered data too early.";
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc