Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.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.
  */
 
@@ skipping 306 matching lines @@
     assert(*frame_size_samples > 0);
     if (!replacement_audio_file->Read(*frame_size_samples,
                                       (*replacement_audio).get())) {
       std::cerr << "Could not read replacement audio file." << std::endl;
       webrtc::Trace::ReturnTrace();
       exit(1);
     }
     // Encode it as PCM16.
     assert((*payload).get());
     payload_len = WebRtcPcm16b_Encode((*replacement_audio).get(),
-                                      static_cast<int16_t>(*frame_size_samples),
+                                      *frame_size_samples,
                                       (*payload).get());
     assert(payload_len == 2 * *frame_size_samples);
     // Change payload type to PCM16.
     switch (CodecSampleRate(rtp_header->header.payloadType)) {
       case 8000:
         rtp_header->header.payloadType = static_cast<uint8_t>(FLAGS_pcm16b);
         break;
       case 16000:
         rtp_header->header.payloadType = static_cast<uint8_t>(FLAGS_pcm16b_wb);
         break;
@@ skipping 13 matching lines @@
         exit(1);
     }
   }
   return payload_len;
 }
 
 }  // namespace
 
 int main(int argc, char* argv[]) {
   static const int kMaxChannels = 5;
-  static const int kMaxSamplesPerMs = 48000 / 1000;
+  static const size_t kMaxSamplesPerMs = 48000 / 1000;
   static const int kOutputBlockSizeMs = 10;
 
   std::string program_name = argv[0];
   std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
       "Run " + program_name + " --helpshort for usage.\n"
       "Example usage:\n" + program_name +
       " input.rtp output.{pcm, wav}\n";
   google::SetUsageMessage(usage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
@@ skipping 173 matching lines @@
         if (packet_available != next_packet_available) {
           packet_available = !packet_available;
           next_packet_available = !next_packet_available;
         }
       }
       next_input_time_ms = packet->time_ms();
     }
 
     // Check if it is time to get output audio.
     if (time_now_ms >= next_output_time_ms) {
-      static const int kOutDataLen =
+      static const size_t kOutDataLen =
           kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
       int16_t out_data[kOutDataLen];
       int num_channels;
-      int samples_per_channel;
+      size_t samples_per_channel;
       int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                    &num_channels, NULL);
       if (error != NetEq::kOK) {
         std::cerr << "GetAudio returned error code " <<
             neteq->LastError() << std::endl;
       } else {
         // Calculate sample rate from output size.
-        sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
+        sample_rate_hz =
+            static_cast<int>(1000 * samples_per_channel / kOutputBlockSizeMs);

[hlundin-webrtc, 2015/08/10 11:30:02]

rtc::checked_cast

[Peter Kasting, 2015/08/17 22:49:47]

Done.

       }
 
       // Write to file.
       // TODO(hlundin): Make writing to file optional.
       size_t write_len = samples_per_channel * num_channels;
       if (!output->WriteArray(out_data, write_len)) {
         std::cerr << "Error while writing to file" << std::endl;
         webrtc::Trace::ReturnTrace();
         exit(1);
       }
       next_output_time_ms += kOutputBlockSizeMs;
     }
     // Advance time to next event.
     time_now_ms = std::min(next_input_time_ms, next_output_time_ms);
   }
 
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n", time_now_ms - start_time_ms);
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }