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-rw-r--r--talk/app/webrtc/webrtcsdp.cc2
-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
-rw-r--r--webrtc/test/field_trial.cc2
-rw-r--r--webrtc/tools/agc/agc_test.cc2
-rw-r--r--webrtc/video/end_to_end_tests.cc2
-rw-r--r--webrtc/video/replay.cc8
12 files changed, 20 insertions, 21 deletions
diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index 9d647955d1..b6f23ca9d0 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -2192,7 +2192,7 @@ bool ParseMediaDescription(const std::string& message,
for (size_t j = 3 ; j < fields.size(); ++j) {
// TODO(wu): Remove when below bug is fixed.
// https://bugzilla.mozilla.org/show_bug.cgi?id=996329
- if (fields[j] == "" && j == fields.size() - 1) {
+ if (fields[j].empty() && j == fields.size() - 1) {
continue;
}
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,
diff --git a/webrtc/test/field_trial.cc b/webrtc/test/field_trial.cc
index 92aa6b0815..05ca052a57 100644
--- a/webrtc/test/field_trial.cc
+++ b/webrtc/test/field_trial.cc
@@ -49,7 +49,7 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
assert(field_trials_initiated_ == false);
field_trials_initiated_ = true;
- if (trials_string == "")
+ if (trials_string.empty())
return;
size_t next_item = 0;
diff --git a/webrtc/tools/agc/agc_test.cc b/webrtc/tools/agc/agc_test.cc
index a651cb168e..e8c4eb8884 100644
--- a/webrtc/tools/agc/agc_test.cc
+++ b/webrtc/tools/agc/agc_test.cc
@@ -91,7 +91,7 @@ void RunAgc() {
ASSERT_TRUE(out_file != NULL);
int gain_map[256];
- if (FLAGS_gain_file != "") {
+ if (!FLAGS_gain_file.empty()) {
FILE* gain_file = fopen(FLAGS_gain_file.c_str(), "rt");
ASSERT_TRUE(gain_file != NULL);
ReadGainMapFromFile(gain_file, FLAGS_gain_offset, gain_map);
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index dd9650db05..82fbe3ae52 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -2001,7 +2001,7 @@ TEST_F(EndToEndTest, GetStats) {
stats.frame_counts.key_frames != 0 ||
stats.frame_counts.delta_frames != 0;
- receive_stats_filled_["CName"] |= stats.c_name != "";
+ receive_stats_filled_["CName"] |= !stats.c_name.empty();
receive_stats_filled_["RtcpPacketTypeCount"] |=
stats.rtcp_packet_type_counts.fir_packets != 0 ||
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index 4b5aa0fb5f..fa61257bbf 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -106,7 +106,7 @@ static const bool timestamp_offset_dummy =
// Flag for rtpdump input file.
bool ValidateInputFilenameNotEmpty(const char* flagname,
const std::string& string) {
- return string != "";
+ return !string.empty();
}
DEFINE_string(input_file, "", "input file");
@@ -156,7 +156,7 @@ class FileRenderPassthrough : public VideoRenderer {
int time_to_render_ms) override {
if (renderer_ != nullptr)
renderer_->RenderFrame(video_frame, time_to_render_ms);
- if (basename_ == "")
+ if (basename_.empty())
return;
if (last_width_ != video_frame.width() ||
last_height_ != video_frame.height()) {
@@ -241,13 +241,13 @@ void RtpReplay() {
encoder_settings.payload_type = flags::PayloadType();
VideoReceiveStream::Decoder decoder;
rtc::scoped_ptr<DecoderBitstreamFileWriter> bitstream_writer;
- if (flags::DecoderBitstreamFilename() != "") {
+ if (!flags::DecoderBitstreamFilename().empty()) {
bitstream_writer.reset(new DecoderBitstreamFileWriter(
flags::DecoderBitstreamFilename().c_str()));
receive_config.pre_decode_callback = bitstream_writer.get();
}
decoder = test::CreateMatchingDecoder(encoder_settings);
- if (flags::DecoderBitstreamFilename() != "") {
+ if (!flags::DecoderBitstreamFilename().empty()) {
// Replace with a null decoder if we're writing the bitstream to a file
// instead.
delete decoder.decoder;
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-11 20:00:32 +0000