[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_processing/test/unpack.cc

 /*
  *  Copyright (c) 2011 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.
  */
 
 // Commandline tool to unpack audioproc debug files.
 //
 // The debug files are dumped as protobuf blobs. For analysis, it's necessary
 // to unpack the file into its component parts: audio and other data.
 
 #include <stdio.h>
 
 #include "gflags/gflags.h"
 #include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/typedefs.h"
 
 // TODO(andrew): unpack more of the data.
 DEFINE_string(input_file, "input", "The name of the input stream file.");
 DEFINE_string(output_file, "ref_out",
               "The name of the reference output stream file.");
 DEFINE_string(reverse_file, "reverse",
@@ skipping 34 matching lines @@
 
   if (argc < 2) {
     printf("%s", google::ProgramUsage());
     return 1;
   }
 
   FILE* debug_file = OpenFile(argv[1], "rb");
 
   Event event_msg;
   int frame_count = 0;
-  int reverse_samples_per_channel = 0;
-  int input_samples_per_channel = 0;
-  int output_samples_per_channel = 0;
-  int num_reverse_channels = 0;
-  int num_input_channels = 0;
-  int num_output_channels = 0;
+  size_t reverse_samples_per_channel = 0;
+  size_t input_samples_per_channel = 0;
+  size_t output_samples_per_channel = 0;
+  size_t num_reverse_channels = 0;
+  size_t num_input_channels = 0;
+  size_t num_output_channels = 0;
   rtc::scoped_ptr<WavWriter> reverse_wav_file;
   rtc::scoped_ptr<WavWriter> input_wav_file;
   rtc::scoped_ptr<WavWriter> output_wav_file;
   rtc::scoped_ptr<RawFile> reverse_raw_file;
   rtc::scoped_ptr<RawFile> input_raw_file;
   rtc::scoped_ptr<RawFile> output_raw_file;
   while (ReadMessageFromFile(debug_file, &event_msg)) {
     if (event_msg.type() == Event::REVERSE_STREAM) {
       if (!event_msg.has_reverse_stream()) {
         printf("Corrupt input file: ReverseStream missing.\n");
@@ skipping 12 matching lines @@
         WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
                      num_reverse_channels * reverse_samples_per_channel,
                      reverse_wav_file.get(),
                      reverse_raw_file.get());
       } else if (msg.channel_size() > 0) {
         if (FLAGS_raw && !reverse_raw_file) {
           reverse_raw_file.reset(new RawFile(FLAGS_reverse_file + ".float"));
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_reverse_channels]);
-        for (int i = 0; i < num_reverse_channels; ++i) {
+        for (size_t i = 0; i < num_reverse_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
         }
         WriteFloatData(data.get(),
                        reverse_samples_per_channel,
                        num_reverse_channels,
                        reverse_wav_file.get(),
                        reverse_raw_file.get());
       }
     } else if (event_msg.type() == Event::STREAM) {
       frame_count++;
@@ skipping 10 matching lines @@
         WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
                      num_input_channels * input_samples_per_channel,
                      input_wav_file.get(),
                      input_raw_file.get());
       } else if (msg.input_channel_size() > 0) {
         if (FLAGS_raw && !input_raw_file) {
           input_raw_file.reset(new RawFile(FLAGS_input_file + ".float"));
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_input_channels]);
-        for (int i = 0; i < num_input_channels; ++i) {
+        for (size_t i = 0; i < num_input_channels; ++i) {
           data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
         }
         WriteFloatData(data.get(),
                        input_samples_per_channel,
                        num_input_channels,
                        input_wav_file.get(),
                        input_raw_file.get());
       }
 
       if (msg.has_output_data()) {
         if (FLAGS_raw && !output_raw_file) {
           output_raw_file.reset(new RawFile(FLAGS_output_file + ".pcm"));
         }
         WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
                      num_output_channels * output_samples_per_channel,
                      output_wav_file.get(),
                      output_raw_file.get());
       } else if (msg.output_channel_size() > 0) {
         if (FLAGS_raw && !output_raw_file) {
           output_raw_file.reset(new RawFile(FLAGS_output_file + ".float"));
         }
         rtc::scoped_ptr<const float* []> data(
             new const float* [num_output_channels]);
-        for (int i = 0; i < num_output_channels; ++i) {
+        for (size_t i = 0; i < num_output_channels; ++i) {
           data[i] =
               reinterpret_cast<const float*>(msg.output_channel(i).data());
         }
         WriteFloatData(data.get(),
                        output_samples_per_channel,
                        num_output_channels,
                        output_wav_file.get(),
                        output_raw_file.get());
       }
 
@@ skipping 51 matching lines @@
       fprintf(settings_file, "Init at frame: %d\n", frame_count);
       int input_sample_rate = msg.sample_rate();
       fprintf(settings_file, "  Input sample rate: %d\n", input_sample_rate);
       int output_sample_rate = msg.output_sample_rate();
       fprintf(settings_file, "  Output sample rate: %d\n", output_sample_rate);
       int reverse_sample_rate = msg.reverse_sample_rate();
       fprintf(settings_file,
               "  Reverse sample rate: %d\n",
               reverse_sample_rate);
       num_input_channels = msg.num_input_channels();
-      fprintf(settings_file, "  Input channels: %d\n", num_input_channels);
+      fprintf(settings_file, "  Input channels: %" PRIuS "\n",
+              num_input_channels);
       num_output_channels = msg.num_output_channels();
-      fprintf(settings_file, "  Output channels: %d\n", num_output_channels);
+      fprintf(settings_file, "  Output channels: %" PRIuS "\n",
+              num_output_channels);
       num_reverse_channels = msg.num_reverse_channels();
-      fprintf(settings_file, "  Reverse channels: %d\n", num_reverse_channels);
+      fprintf(settings_file, "  Reverse channels: %" PRIuS "\n",
+              num_reverse_channels);
 
       fprintf(settings_file, "\n");
 
       if (reverse_sample_rate == 0) {
         reverse_sample_rate = input_sample_rate;
       }
       if (output_sample_rate == 0) {
         output_sample_rate = input_sample_rate;
       }
 
-      reverse_samples_per_channel = reverse_sample_rate / 100;
-      input_samples_per_channel = input_sample_rate / 100;
-      output_samples_per_channel = output_sample_rate / 100;
+      reverse_samples_per_channel =
+          static_cast<size_t>(reverse_sample_rate / 100);
+      input_samples_per_channel =
+          static_cast<size_t>(input_sample_rate / 100);
+      output_samples_per_channel =
+          static_cast<size_t>(output_sample_rate / 100);
 
       if (!FLAGS_raw) {
         // The WAV files need to be reset every time, because they cant change
         // their sample rate or number of channels.
         reverse_wav_file.reset(new WavWriter(FLAGS_reverse_file + ".wav",
                                              reverse_sample_rate,
                                              num_reverse_channels));
         input_wav_file.reset(new WavWriter(FLAGS_input_file + ".wav",
                                            input_sample_rate,
                                            num_input_channels));
         output_wav_file.reset(new WavWriter(FLAGS_output_file + ".wav",
                                             output_sample_rate,
                                             num_output_channels));
       }
     }
   }
 
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::do_main(argc, argv);
 }