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Unified Diff: webrtc/modules/audio_processing/test/unpack.cc

Issue 1238083005: [NOT FOR REVIEW] Convert channel counts to size_t. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@size_t
Patch Set: Checkpoint Created 5 years, 5 months ago
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Index: webrtc/modules/audio_processing/test/unpack.cc
diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
index 2484828bd18662ea1bb5ae15306dd0adcd28719f..872dafa7c2269645de327587c32da95430dc87d1 100644
--- a/webrtc/modules/audio_processing/test/unpack.cc
+++ b/webrtc/modules/audio_processing/test/unpack.cc
@@ -17,6 +17,7 @@
#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"
@@ -71,12 +72,12 @@ int do_main(int argc, char* argv[]) {
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;
@@ -109,7 +110,7 @@ int do_main(int argc, char* argv[]) {
}
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(),
@@ -140,7 +141,7 @@ int do_main(int argc, char* argv[]) {
}
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(),
@@ -164,7 +165,7 @@ int do_main(int argc, char* argv[]) {
}
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());
}
@@ -236,11 +237,14 @@ int do_main(int argc, char* argv[]) {
" 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");
@@ -251,9 +255,12 @@ int do_main(int argc, char* argv[]) {
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
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