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authorpbos <pbos@webrtc.org>2015-07-09 07:48:14 -0700
committerCommit bot <commit-bot@chromium.org>2015-07-09 14:48:27 +0000
commitbb36fdf95f9667fb1f3fbf3073bd15007681322c (patch)
treeb4c0d85b61cfddf1f729908157232f7f86f6ba5d /webrtc/modules
parent3b1e647b6a6f74d8e4392e012fe2262b3d2c4334 (diff)
downloadwebrtc-bb36fdf95f9667fb1f3fbf3073bd15007681322c.tar.gz
Remove empty-string comparisons.
Use .empty() and !.empty() in favor of == "" or != "". BUG= R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1228913003 Cr-Commit-Position: refs/heads/master@{#9559}
Diffstat (limited to 'webrtc/modules')
-rw-r--r--webrtc/modules/audio_processing/test/audio_processing_unittest.cc2
-rw-r--r--webrtc/modules/audio_processing/test/audioproc_float.cc6
-rw-r--r--webrtc/modules/audio_processing/transient/transient_suppression_test.cc4
-rw-r--r--webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc4
-rw-r--r--webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc4
-rw-r--r--webrtc/modules/video_coding/main/test/test_util.cc2
-rw-r--r--webrtc/modules/video_coding/main/test/video_rtp_play.cc3
7 files changed, 12 insertions, 13 deletions
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index c7c45f7eeb..291035a012 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -269,7 +269,7 @@ std::string OutputFilePath(std::string name,
ss << output_rate / 1000 << "_pcm";
std::string filename = ss.str();
- if (temp_filenames[filename] == "")
+ if (temp_filenames[filename].empty())
temp_filenames[filename] = test::TempFilename(test::OutputPath(), filename);
return temp_filenames[filename];
}
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 381d7fd2b6..dac43629cf 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -64,12 +64,12 @@ int main(int argc, char* argv[]) {
google::SetUsageMessage(kUsage);
google::ParseCommandLineFlags(&argc, &argv, true);
- if (!((FLAGS_i == "") ^ (FLAGS_dump == ""))) {
+ if (!((FLAGS_i.empty()) ^ (FLAGS_dump.empty()))) {
fprintf(stderr,
"An input file must be specified with either -i or -dump.\n");
return 1;
}
- if (FLAGS_dump != "") {
+ if (!FLAGS_dump.empty()) {
fprintf(stderr, "FIXME: the -dump option is not yet implemented.\n");
return 1;
}
@@ -96,7 +96,7 @@ int main(int argc, char* argv[]) {
}
rtc::scoped_ptr<AudioProcessing> ap(AudioProcessing::Create(config));
- if (FLAGS_dump != "") {
+ if (!FLAGS_dump.empty()) {
CHECK_EQ(kNoErr, ap->echo_cancellation()->Enable(FLAGS_aec || FLAGS_all));
} else if (FLAGS_aec) {
fprintf(stderr, "-aec requires a -dump file.\n");
diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index fdc5686763..506abaf203 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -150,13 +150,13 @@ void void_main() {
// Prepare the detection file.
FILE* detection_file = NULL;
- if (FLAGS_detection_file_name != "") {
+ if (!FLAGS_detection_file_name.empty()) {
detection_file = fopen(FLAGS_detection_file_name.c_str(), "rb");
}
// Prepare the reference file.
FILE* reference_file = NULL;
- if (FLAGS_reference_file_name != "") {
+ if (!FLAGS_reference_file_name.empty()) {
reference_file = fopen(FLAGS_reference_file_name.c_str(), "rb");
}
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
index 319a97bfca..de13023b26 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
@@ -116,9 +116,9 @@ Logging::State::State(const std::string& tag, int64_t timestamp_ms,
}
void Logging::State::MergePrevious(const State& previous) {
- if (tag == "") {
+ if (tag.empty()) {
tag = previous.tag;
- } else if (previous.tag != "") {
+ } else if (!previous.tag.empty()) {
tag = previous.tag + "_" + tag;
}
timestamp_ms = std::max(previous.timestamp_ms, timestamp_ms);
diff --git a/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc b/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
index ced92bce24..f6eee39f3b 100644
--- a/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
+++ b/webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
@@ -111,7 +111,7 @@ int Log(const char *format, ...) {
// Returns 0 if everything is OK, otherwise an exit code.
int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
// Validate the mandatory flags:
- if (FLAGS_input_filename == "" || FLAGS_width == -1 || FLAGS_height == -1) {
+ if (FLAGS_input_filename.empty() || FLAGS_width == -1 || FLAGS_height == -1) {
printf("%s\n", google::ProgramUsage());
return 1;
}
@@ -140,7 +140,7 @@ int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
config->output_dir = FLAGS_output_dir;
// Manufacture an output filename if none was given.
- if (FLAGS_output_filename == "") {
+ if (FLAGS_output_filename.empty()) {
// Cut out the filename without extension from the given input file
// (which may include a path)
int startIndex = FLAGS_input_filename.find_last_of("/") + 1;
diff --git a/webrtc/modules/video_coding/main/test/test_util.cc b/webrtc/modules/video_coding/main/test/test_util.cc
index ce71cbb49d..cd858da288 100644
--- a/webrtc/modules/video_coding/main/test/test_util.cc
+++ b/webrtc/modules/video_coding/main/test/test_util.cc
@@ -73,7 +73,7 @@ FileOutputFrameReceiver::FileOutputFrameReceiver(
count_(0) {
std::string basename;
std::string extension;
- if (base_out_filename == "") {
+ if (base_out_filename.empty()) {
basename = webrtc::test::OutputPath() + "rtp_decoded";
extension = "yuv";
} else {
diff --git a/webrtc/modules/video_coding/main/test/video_rtp_play.cc b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
index c1bc02fbed..256f508327 100644
--- a/webrtc/modules/video_coding/main/test/video_rtp_play.cc
+++ b/webrtc/modules/video_coding/main/test/video_rtp_play.cc
@@ -44,9 +44,8 @@ int RtpPlay(const CmdArgs& args) {
kDefaultVp8PayloadType, "VP8", webrtc::kVideoCodecVP8));
std::string output_file = args.outputFile;
- if (output_file == "") {
+ if (output_file.empty())
output_file = webrtc::test::OutputPath() + "RtpPlay_decoded.yuv";
- }
webrtc::SimulatedClock clock(0);
webrtc::rtpplayer::VcmPayloadSinkFactory factory(output_file, &clock,