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Unified Diff: webrtc/modules/audio_processing/transient/transient_suppression_test.cc

Issue 3005483002: Replace remaining gflags usages with rtc_base/flags (Closed)
Patch Set: Rebase Created 3 years, 4 months ago
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Index: webrtc/modules/audio_processing/transient/transient_suppression_test.cc
diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index 2158a8115893ac4ff621d06309f3562fdc15d694..ea894bc12d32af6dc747912f14799dda3ba97b5a 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -12,14 +12,15 @@
#include <stdlib.h>
#include <stdio.h>
+#include <string.h>
#include <memory>
#include <string>
-#include "gflags/gflags.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/rtc_base/flags.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
@@ -32,31 +33,20 @@ DEFINE_string(reference_file_name,
"",
"PCM file that contains the reference signal.");
-static bool ValidatePositiveInt(const char* flagname, int32_t value) {
- if (value <= 0) {
- printf("%s must be a positive integer.\n", flagname);
- return false;
- }
- return true;
-}
-DEFINE_int32(chunk_size_ms,
- 10,
- "Time between each chunk of samples in milliseconds.");
-static const bool chunk_size_ms_dummy =
- google::RegisterFlagValidator(&FLAGS_chunk_size_ms, &ValidatePositiveInt);
-
-DEFINE_int32(sample_rate_hz,
- 16000,
- "Sampling frequency of the signal in Hertz.");
-static const bool sample_rate_hz_dummy =
- google::RegisterFlagValidator(&FLAGS_sample_rate_hz, &ValidatePositiveInt);
-DEFINE_int32(detection_rate_hz,
- 0,
- "Sampling frequency of the detection signal in Hertz.");
-
-DEFINE_int32(num_channels, 1, "Number of channels.");
-static const bool num_channels_dummy =
- google::RegisterFlagValidator(&FLAGS_num_channels, &ValidatePositiveInt);
+DEFINE_int(chunk_size_ms,
+ 10,
+ "Time between each chunk of samples in milliseconds.");
+
+DEFINE_int(sample_rate_hz,
+ 16000,
+ "Sampling frequency of the signal in Hertz.");
+DEFINE_int(detection_rate_hz,
+ 0,
+ "Sampling frequency of the detection signal in Hertz.");
+
+DEFINE_int(num_channels, 1, "Number of channels.");
+
+DEFINE_bool(help, false, "Print this message.");
namespace webrtc {
@@ -146,19 +136,19 @@ static void WritePCM(FILE* f,
void void_main() {
// TODO(aluebs): Remove all FileWrappers.
// Prepare the input file.
- FILE* in_file = fopen(FLAGS_in_file_name.c_str(), "rb");
+ FILE* in_file = fopen(FLAG_in_file_name, "rb");
ASSERT_TRUE(in_file != NULL);
// Prepare the detection file.
FILE* detection_file = NULL;
- if (!FLAGS_detection_file_name.empty()) {
- detection_file = fopen(FLAGS_detection_file_name.c_str(), "rb");
+ if (strlen(FLAG_detection_file_name) > 0) {
+ detection_file = fopen(FLAG_detection_file_name, "rb");
}
// Prepare the reference file.
FILE* reference_file = NULL;
- if (!FLAGS_reference_file_name.empty()) {
- reference_file = fopen(FLAGS_reference_file_name.c_str(), "rb");
+ if (strlen(FLAG_reference_file_name) > 0) {
+ reference_file = fopen(FLAG_reference_file_name, "rb");
}
// Prepare the output file.
@@ -166,27 +156,27 @@ void void_main() {
FILE* out_file = fopen(out_file_name.c_str(), "wb");
ASSERT_TRUE(out_file != NULL);
- int detection_rate_hz = FLAGS_detection_rate_hz;
+ int detection_rate_hz = FLAG_detection_rate_hz;
if (detection_rate_hz == 0) {
- detection_rate_hz = FLAGS_sample_rate_hz;
+ detection_rate_hz = FLAG_sample_rate_hz;
}
Agc agc;
TransientSuppressor suppressor;
suppressor.Initialize(
- FLAGS_sample_rate_hz, detection_rate_hz, FLAGS_num_channels);
+ FLAG_sample_rate_hz, detection_rate_hz, FLAG_num_channels);
const size_t audio_buffer_size =
- FLAGS_chunk_size_ms * FLAGS_sample_rate_hz / 1000;
+ FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
const size_t detection_buffer_size =
- FLAGS_chunk_size_ms * detection_rate_hz / 1000;
+ FLAG_chunk_size_ms * detection_rate_hz / 1000;
// int16 and float variants of the same data.
std::unique_ptr<int16_t[]> audio_buffer_i(
- new int16_t[FLAGS_num_channels * audio_buffer_size]);
+ new int16_t[FLAG_num_channels * audio_buffer_size]);
std::unique_ptr<float[]> audio_buffer_f(
- new float[FLAGS_num_channels * audio_buffer_size]);
+ new float[FLAG_num_channels * audio_buffer_size]);
std::unique_ptr<float[]> detection_buffer, reference_buffer;
@@ -197,7 +187,7 @@ void void_main() {
while (ReadBuffers(in_file,
audio_buffer_size,
- FLAGS_num_channels,
+ FLAG_num_channels,
audio_buffer_i.get(),
detection_file,
detection_buffer_size,
@@ -207,17 +197,17 @@ void void_main() {
ASSERT_EQ(0,
agc.Process(audio_buffer_i.get(),
static_cast<int>(audio_buffer_size),
- FLAGS_sample_rate_hz))
+ FLAG_sample_rate_hz))
<< "The AGC could not process the frame";
- for (size_t i = 0; i < FLAGS_num_channels * audio_buffer_size; ++i) {
+ for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
audio_buffer_f[i] = audio_buffer_i[i];
}
ASSERT_EQ(0,
suppressor.Suppress(audio_buffer_f.get(),
audio_buffer_size,
- FLAGS_num_channels,
+ FLAG_num_channels,
detection_buffer.get(),
detection_buffer_size,
reference_buffer.get(),
@@ -228,7 +218,7 @@ void void_main() {
// Write result to out file.
WritePCM(
- out_file, audio_buffer_size, FLAGS_num_channels, audio_buffer_f.get());
+ out_file, audio_buffer_size, FLAG_num_channels, audio_buffer_f.get());
}
fclose(in_file);
@@ -244,8 +234,19 @@ void void_main() {
} // namespace webrtc
int main(int argc, char* argv[]) {
- google::SetUsageMessage(webrtc::kUsage);
- google::ParseCommandLineFlags(&argc, &argv, true);
+ if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
+ FLAG_help || argc != 1) {
+ printf("%s", webrtc::kUsage);
+ if (FLAG_help) {
+ rtc::FlagList::Print(nullptr, false);
+ return 0;
+ }
+ return 1;
+ }
+ RTC_CHECK_GT(FLAG_chunk_size_ms, 0);
+ RTC_CHECK_GT(FLAG_sample_rate_hz, 0);
+ RTC_CHECK_GT(FLAG_num_channels, 0);
+
webrtc::void_main();
return 0;
}
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