Prepare to convert various types to size_t.

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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 70 matching lines @@
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
   for (int i = 0; i < num_samples; ++i) {
     speech_buffer_.push_back(audio[i]);
   }
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
   CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
   CHECK_GE(max_encoded_bytes, full_frame_samples_);
-  int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
-  CHECK_GE(ret, 0);
-  speech_buffer_.clear();
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
+  int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
+  CHECK_GE(ret, 0);
   info.encoded_bytes = static_cast<size_t>(ret);
+  speech_buffer_.clear();
   return info;
 }
 
 int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
                                      size_t input_len,
                                      uint8_t* encoded) {
   return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded);
 }
 
 int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
@@ skipping 17 matching lines @@
     : AudioEncoderMutableImpl<AudioEncoderPcmU>(
           CreateConfig<AudioEncoderPcmU>(codec_inst)) {
 }
 
 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmA>(
           CreateConfig<AudioEncoderPcmA>(codec_inst)) {
 }
 
 }  // namespace webrtc