| 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
|
|
|