RTC_[D]CHECK_op: Remove "u" suffix on integer constants
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc
index 163cc11..7036c10 100644
--- a/webrtc/audio/audio_transport_proxy.cc
+++ b/webrtc/audio/audio_transport_proxy.cc
@@ -68,8 +68,8 @@
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
- RTC_DCHECK_GE(nChannels, 1u);
- RTC_DCHECK_LE(nChannels, 2u);
+ RTC_DCHECK_GE(nChannels, 1);
+ RTC_DCHECK_LE(nChannels, 2);
RTC_DCHECK_GE(
samplesPerSec,
static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
@@ -111,8 +111,8 @@
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) {
RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 16);
- RTC_DCHECK_GE(number_of_channels, 1u);
- RTC_DCHECK_LE(number_of_channels, 2u);
+ RTC_DCHECK_GE(number_of_channels, 1);
+ RTC_DCHECK_LE(number_of_channels, 2);
RTC_DCHECK_GE(static_cast<int>(sample_rate),
AudioProcessing::NativeRate::kSampleRate8kHz);
diff --git a/webrtc/base/bitbuffer.cc b/webrtc/base/bitbuffer.cc
index 48a1d0c..fc8a899 100644
--- a/webrtc/base/bitbuffer.cc
+++ b/webrtc/base/bitbuffer.cc
@@ -19,14 +19,14 @@
// Returns the lowest (right-most) |bit_count| bits in |byte|.
uint8_t LowestBits(uint8_t byte, size_t bit_count) {
- RTC_DCHECK_LE(bit_count, 8u);
+ RTC_DCHECK_LE(bit_count, 8);
return byte & ((1 << bit_count) - 1);
}
// Returns the highest (left-most) |bit_count| bits in |byte|, shifted to the
// lowest bits (to the right).
uint8_t HighestBits(uint8_t byte, size_t bit_count) {
- RTC_DCHECK_LE(bit_count, 8u);
+ RTC_DCHECK_LE(bit_count, 8);
uint8_t shift = 8 - static_cast<uint8_t>(bit_count);
uint8_t mask = 0xFF << shift;
return (byte & mask) >> shift;
diff --git a/webrtc/base/filerotatingstream.cc b/webrtc/base/filerotatingstream.cc
index 0809994..d1434de 100644
--- a/webrtc/base/filerotatingstream.cc
+++ b/webrtc/base/filerotatingstream.cc
@@ -37,8 +37,8 @@
max_file_size,
num_files,
kWrite) {
- RTC_DCHECK_GT(max_file_size, 0u);
- RTC_DCHECK_GT(num_files, 1u);
+ RTC_DCHECK_GT(max_file_size, 0);
+ RTC_DCHECK_GT(num_files, 1);
}
FileRotatingStream::FileRotatingStream(const std::string& dir_path,
@@ -248,7 +248,7 @@
case kWrite:
mode = "w+";
// We should always we writing to the zero-th file.
- RTC_DCHECK_EQ(current_file_index_, 0u);
+ RTC_DCHECK_EQ(current_file_index_, 0);
break;
case kRead:
mode = "r";
@@ -360,7 +360,7 @@
GetNumRotatingLogFiles(max_total_log_size) + 1),
max_total_log_size_(max_total_log_size),
num_rotations_(0) {
- RTC_DCHECK_GE(max_total_log_size, 4u);
+ RTC_DCHECK_GE(max_total_log_size, 4);
}
const char* CallSessionFileRotatingStream::kLogPrefix = "webrtc_log";
diff --git a/webrtc/base/flags.cc b/webrtc/base/flags.cc
index a138b8f..4fcd4ac 100644
--- a/webrtc/base/flags.cc
+++ b/webrtc/base/flags.cc
@@ -258,7 +258,7 @@
void FlagList::Register(Flag* flag) {
RTC_DCHECK(flag);
- RTC_DCHECK_GT(strlen(flag->name()), 0u);
+ RTC_DCHECK_GT(strlen(flag->name()), 0);
// NOTE: Don't call Lookup() within Register because it accesses the name_
// of other flags in list_, and if the flags are coming from two different
// compilation units, the initialization order between them is undefined, and
diff --git a/webrtc/call/bitrate_allocator.cc b/webrtc/call/bitrate_allocator.cc
index 2c5ba6d..6e9be73 100644
--- a/webrtc/call/bitrate_allocator.cc
+++ b/webrtc/call/bitrate_allocator.cc
@@ -36,7 +36,7 @@
namespace {
double MediaRatio(uint32_t allocated_bitrate, uint32_t protection_bitrate) {
- RTC_DCHECK_GT(allocated_bitrate, 0u);
+ RTC_DCHECK_GT(allocated_bitrate, 0);
if (protection_bitrate == 0)
return 1.0;
@@ -382,7 +382,7 @@
}
auto it = list_max_bitrates.begin();
while (it != list_max_bitrates.end()) {
- RTC_DCHECK_GT(bitrate, 0u);
+ RTC_DCHECK_GT(bitrate, 0);
uint32_t extra_allocation =
bitrate / static_cast<uint32_t>(list_max_bitrates.size());
uint32_t total_allocation =
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index ddb69ed..95d42ef 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -166,7 +166,7 @@
send_stream_ = test_->sender_call_->CreateVideoSendStream(
test_->video_send_config_.Copy(),
test_->video_encoder_config_.Copy());
- RTC_DCHECK_EQ(1u, test_->video_encoder_config_.number_of_streams);
+ RTC_DCHECK_EQ(1, test_->video_encoder_config_.number_of_streams);
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
kDefaultWidth, kDefaultHeight, kDefaultFramerate,
Clock::GetRealTimeClock()));
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 354e092..fb5ae0d 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -549,7 +549,7 @@
Action OnSendRtp(const uint8_t* packet, size_t length) override {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (stats.substreams.size() > 0) {
- RTC_DCHECK_EQ(1u, stats.substreams.size());
+ RTC_DCHECK_EQ(1, stats.substreams.size());
int bitrate_kbps =
stats.substreams.begin()->second.total_bitrate_bps / 1000;
if (bitrate_kbps > min_acceptable_bitrate_ &&
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index d1dcacd..231ba88 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -109,7 +109,7 @@
public:
CompositionConverter(std::vector<std::unique_ptr<AudioConverter>> converters)
: converters_(std::move(converters)) {
- RTC_CHECK_GE(converters_.size(), 2u);
+ RTC_CHECK_GE(converters_.size(), 2);
// We need an intermediate buffer after every converter.
for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
buffers_.push_back(
diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h
index f4ed683..c4c85d1 100644
--- a/webrtc/common_audio/channel_buffer.h
+++ b/webrtc/common_audio/channel_buffer.h
@@ -94,7 +94,7 @@
// 0 <= sample < |num_frames_per_band_|
const T* const* bands(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
- RTC_DCHECK_GE(channel, 0u);
+ RTC_DCHECK_GE(channel, 0);
return &bands_[channel * num_bands_];
}
T* const* bands(size_t channel) {
diff --git a/webrtc/common_audio/include/audio_util.h b/webrtc/common_audio/include/audio_util.h
index e5ad701..1601c7f 100644
--- a/webrtc/common_audio/include/audio_util.h
+++ b/webrtc/common_audio/include/audio_util.h
@@ -154,7 +154,7 @@
int num_channels,
T* deinterleaved) {
RTC_DCHECK_GT(num_channels, 0);
- RTC_DCHECK_GT(num_frames, 0u);
+ RTC_DCHECK_GT(num_frames, 0);
const T* const end = interleaved + num_frames * num_channels;
diff --git a/webrtc/common_audio/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc
index 8a791f3..6825beb 100644
--- a/webrtc/common_audio/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -84,12 +84,12 @@
cplx_length_,
RealFourier::kFftBufferAlignment) {
RTC_CHECK(num_in_channels_ > 0);
- RTC_CHECK_GT(block_length_, 0u);
- RTC_CHECK_GT(chunk_length_, 0u);
+ RTC_CHECK_GT(block_length_, 0);
+ RTC_CHECK_GT(chunk_length_, 0);
RTC_CHECK(block_processor_);
// block_length_ power of 2?
- RTC_CHECK_EQ(0u, block_length_ & (block_length_ - 1));
+ RTC_CHECK_EQ(0, block_length_ & (block_length_ - 1));
}
LappedTransform::~LappedTransform() = default;
diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc
index 9f329c4..788223d 100644
--- a/webrtc/common_audio/resampler/push_resampler.cc
+++ b/webrtc/common_audio/resampler/push_resampler.cc
@@ -32,8 +32,8 @@
#if !defined(WEBRTC_WIN) && defined(__clang__)
RTC_DCHECK_GT(src_sample_rate_hz, 0);
RTC_DCHECK_GT(dst_sample_rate_hz, 0);
- RTC_DCHECK_GT(num_channels, 0u);
- RTC_DCHECK_LE(num_channels, 2u);
+ RTC_DCHECK_GT(num_channels, 0);
+ RTC_DCHECK_LE(num_channels, 2);
#endif
}
diff --git a/webrtc/common_audio/sparse_fir_filter.cc b/webrtc/common_audio/sparse_fir_filter.cc
index a79da07..2928004 100644
--- a/webrtc/common_audio/sparse_fir_filter.cc
+++ b/webrtc/common_audio/sparse_fir_filter.cc
@@ -22,8 +22,8 @@
offset_(offset),
nonzero_coeffs_(nonzero_coeffs, nonzero_coeffs + num_nonzero_coeffs),
state_(sparsity_ * (num_nonzero_coeffs - 1) + offset_, 0.f) {
- RTC_CHECK_GE(num_nonzero_coeffs, 1u);
- RTC_CHECK_GE(sparsity, 1u);
+ RTC_CHECK_GE(num_nonzero_coeffs, 1);
+ RTC_CHECK_GE(sparsity, 1);
}
SparseFIRFilter::~SparseFIRFilter() = default;
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index 572c94b..2b9098a 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -123,7 +123,7 @@
// Write a blank placeholder header, since we need to know the total number
// of samples before we can fill in the real data.
static const uint8_t blank_header[kWavHeaderSize] = {0};
- RTC_CHECK_EQ(1u, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
+ RTC_CHECK_EQ(1, fwrite(blank_header, kWavHeaderSize, 1, file_handle_));
}
WavWriter::~WavWriter() {
@@ -168,7 +168,7 @@
uint8_t header[kWavHeaderSize];
WriteWavHeader(header, num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_);
- RTC_CHECK_EQ(1u, fwrite(header, kWavHeaderSize, 1, file_handle_));
+ RTC_CHECK_EQ(1, fwrite(header, kWavHeaderSize, 1, file_handle_));
RTC_CHECK_EQ(0, fclose(file_handle_));
file_handle_ = NULL;
}
diff --git a/webrtc/media/engine/webrtcvideoengine2.cc b/webrtc/media/engine/webrtcvideoengine2.cc
index e947fd1..3f9d5f6 100644
--- a/webrtc/media/engine/webrtcvideoengine2.cc
+++ b/webrtc/media/engine/webrtcvideoengine2.cc
@@ -1780,7 +1780,7 @@
const VideoCodecSettings& codec_settings) {
RTC_DCHECK_RUN_ON(&thread_checker_);
parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
- RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u);
+ RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
parameters_.config.encoder_settings.encoder = new_encoder.encoder;
@@ -1961,7 +1961,7 @@
return;
}
- RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0u);
+ RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
RTC_CHECK(parameters_.codec_settings);
VideoCodecSettings codec_settings = *parameters_.codec_settings;
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index ee1034b..dd02964 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -536,7 +536,7 @@
frame_type = kEmptyFrame;
encoded_info.payload_type = previous_pltype;
} else {
- RTC_DCHECK_GT(encode_buffer_.size(), 0u);
+ RTC_DCHECK_GT(encode_buffer_.size(), 0);
frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
}
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc
index d8dcd79..afeefc7 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc
@@ -211,7 +211,7 @@
if (sub_enc.empty()) {
break;
}
- RTC_CHECK_EQ(1u, sub_enc.size());
+ RTC_CHECK_EQ(1, sub_enc.size());
// Replace enc with its sub encoder. We need to put the sub encoder in
// a temporary first, since otherwise the old value of enc would be
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index d2edcb5..b162941 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -240,9 +240,9 @@
samples_per_10ms_frame),
encoded);
if (i + 1 == frames_to_encode) {
- RTC_CHECK_GT(info.encoded_bytes, 0u) << "Encoder didn't deliver data.";
+ RTC_CHECK_GT(info.encoded_bytes, 0) << "Encoder didn't deliver data.";
} else {
- RTC_CHECK_EQ(info.encoded_bytes, 0u)
+ RTC_CHECK_EQ(info.encoded_bytes, 0)
<< "Encoder delivered data too early.";
}
}
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 483311b..b1d259e 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -20,7 +20,7 @@
class AudioDecoderPcmU final : public AudioDecoder {
public:
explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
- RTC_DCHECK_GE(num_channels, 1u);
+ RTC_DCHECK_GE(num_channels, 1);
}
void Reset() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
@@ -44,7 +44,7 @@
class AudioDecoderPcmA final : public AudioDecoder {
public:
explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
- RTC_DCHECK_GE(num_channels, 1u);
+ RTC_DCHECK_GE(num_channels, 1);
}
void Reset() override;
std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
index 354f819..a0fc02b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -76,7 +76,7 @@
return results;
}
- RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame);
+ RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
if (payload.size() == bytes_per_frame) {
std::unique_ptr<EncodedAudioFrame> frame(
new LegacyEncodedAudioFrame(this, std::move(payload)));
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index e9772f6..1dc312f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -452,7 +452,7 @@
}
void AudioEncoderOpus::SetNumChannelsToEncode(size_t num_channels_to_encode) {
- RTC_DCHECK_GT(num_channels_to_encode, 0u);
+ RTC_DCHECK_GT(num_channels_to_encode, 0);
RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
if (num_channels_to_encode_ == num_channels_to_encode)
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 54529ad..8e59f49 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -180,7 +180,7 @@
// Returns a vector with the n evenly-spaced numbers a, a + (b - a)/(n - 1),
// ..., b.
std::vector<double> IntervalSteps(double a, double b, size_t n) {
- RTC_DCHECK_GT(n, 1u);
+ RTC_DCHECK_GT(n, 1);
const double step = (b - a) / (n - 1);
std::vector<double> points;
for (size_t i = 0; i < n; ++i)
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 43d2dac..692e212 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -21,7 +21,7 @@
RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
sample_rate_hz == 32000 || sample_rate_hz == 48000)
<< "Unsupported sample rate " << sample_rate_hz;
- RTC_DCHECK_GE(num_channels, 1u);
+ RTC_DCHECK_GE(num_channels, 1);
}
void AudioDecoderPcm16B::Reset() {}
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 37fa55a..07db78c 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -71,11 +71,11 @@
// discarding the (empty) vector of redundant information. This is
// intentional.
info.redundant.push_back(info);
- RTC_DCHECK_EQ(info.redundant.size(), 1u);
+ RTC_DCHECK_EQ(info.redundant.size(), 1);
if (secondary_info_.encoded_bytes > 0) {
encoded->AppendData(secondary_encoded_);
info.redundant.push_back(secondary_info_);
- RTC_DCHECK_EQ(info.redundant.size(), 2u);
+ RTC_DCHECK_EQ(info.redundant.size(), 2);
}
// Save primary to secondary.
secondary_encoded_.SetData(encoded->data() + primary_offset,
diff --git a/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc b/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
index 10535e2..5763572 100644
--- a/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
@@ -66,7 +66,7 @@
if (max_period_element == peak_history_.end()) {
return 0; // |peak_history_| is empty.
}
- RTC_DCHECK_GT(max_period_element->period_ms, 0u);
+ RTC_DCHECK_GT(max_period_element->period_ms, 0);
return max_period_element->period_ms;
}
diff --git a/webrtc/modules/audio_coding/neteq/nack_tracker.cc b/webrtc/modules/audio_coding/neteq/nack_tracker.cc
index 62cb2ec..f978793 100644
--- a/webrtc/modules/audio_coding/neteq/nack_tracker.cc
+++ b/webrtc/modules/audio_coding/neteq/nack_tracker.cc
@@ -196,7 +196,7 @@
}
void NackTracker::SetMaxNackListSize(size_t max_nack_list_size) {
- RTC_CHECK_GT(max_nack_list_size, 0u);
+ RTC_CHECK_GT(max_nack_list_size, 0);
// Ugly hack to get around the problem of passing static consts by reference.
const size_t kNackListSizeLimitLocal = NackTracker::kNackListSizeLimit;
RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 4ccf4fb..f390f53 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -1724,7 +1724,7 @@
#ifdef CODEC_OPUS
cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12,
encoded);
- RTC_CHECK_GT(cdlen, 0u);
+ RTC_CHECK_GT(cdlen, 0);
#endif
indata += frameLen;
encoded += cdlen;
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index c89200b..263f7b4 100644
--- a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -59,7 +59,7 @@
// Create a new PacketData object.
RTC_DCHECK(!packet_data_);
packet_data_.reset(new NetEqInput::PacketData);
- RTC_DCHECK_EQ(packet_data_->payload.size(), 0u);
+ RTC_DCHECK_EQ(packet_data_->payload.size(), 0);
// Loop until we get a packet.
AudioEncoder::EncodedInfo info;
diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index 29beed5..2e452e1 100644
--- a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -26,13 +26,13 @@
// Decoder is asked to produce codec-internal comfort noise.
RTC_DCHECK(!encoded); // NetEq always sends nullptr in this case.
RTC_DCHECK(cng_mode_);
- RTC_DCHECK_GT(last_decoded_length_, 0u);
+ RTC_DCHECK_GT(last_decoded_length_, 0);
std::fill_n(decoded, last_decoded_length_, 0);
*speech_type = kComfortNoise;
return last_decoded_length_;
}
- RTC_CHECK_GE(encoded_len, 12u);
+ RTC_CHECK_GE(encoded_len, 12);
uint32_t timestamp_to_decode =
ByteReader<uint32_t>::ReadLittleEndian(encoded);
uint32_t samples_to_decode =
@@ -66,7 +66,7 @@
ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
if (original_payload_size_bytes == 1) {
// This is a comfort noise payload.
- RTC_DCHECK_GT(last_decoded_length_, 0u);
+ RTC_DCHECK_GT(last_decoded_length_, 0);
std::fill_n(decoded, last_decoded_length_, 0);
*speech_type = kComfortNoise;
cng_mode_ = true;
@@ -90,7 +90,7 @@
size_t samples,
size_t original_payload_size_bytes,
rtc::ArrayView<uint8_t> encoded) {
- RTC_CHECK_GE(encoded.size(), 12u);
+ RTC_CHECK_GE(encoded.size(), 12);
ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
rtc::checked_cast<uint32_t>(samples));
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index c9163a1..dd10649 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -70,7 +70,7 @@
RTC_DCHECK(packet_);
- RTC_CHECK_EQ(forbidden_types_.count(packet_->header.header.payloadType), 0u)
+ RTC_CHECK_EQ(forbidden_types_.count(packet_->header.header.payloadType), 0)
<< "Payload type " << static_cast<int>(packet_->header.header.payloadType)
<< " is forbidden.";
diff --git a/webrtc/modules/audio_device/android/opensles_player.cc b/webrtc/modules/audio_device/android/opensles_player.cc
index d675d63..7dfc5ec 100644
--- a/webrtc/modules/audio_device/android/opensles_player.cc
+++ b/webrtc/modules/audio_device/android/opensles_player.cc
@@ -145,8 +145,8 @@
// Verify that the buffer queue is in fact cleared as it should.
SLAndroidSimpleBufferQueueState buffer_queue_state;
(*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
- RTC_DCHECK_EQ(0u, buffer_queue_state.count);
- RTC_DCHECK_EQ(0u, buffer_queue_state.index);
+ RTC_DCHECK_EQ(0, buffer_queue_state.count);
+ RTC_DCHECK_EQ(0, buffer_queue_state.index);
#endif
// The number of lower latency audio players is limited, hence we create the
// audio player in Start() and destroy it in Stop().
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index 8310917..e7a5da7 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -409,7 +409,7 @@
int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
RTC_DCHECK_RUN_ON(&playout_thread_checker_);
- RTC_DCHECK_GT(play_buffer_.size(), 0u);
+ RTC_DCHECK_GT(play_buffer_.size(), 0);
const size_t bytes_per_sample = sizeof(int16_t);
memcpy(audio_buffer, play_buffer_.data(),
play_buffer_.size() * bytes_per_sample);
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index ed8dbad..bf6cac7 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -407,9 +407,9 @@
UInt32 num_frames,
AudioBufferList* io_data) {
// Verify 16-bit, noninterleaved mono PCM signal format.
- RTC_DCHECK_EQ(1u, io_data->mNumberBuffers);
+ RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
AudioBuffer* audio_buffer = &io_data->mBuffers[0];
- RTC_DCHECK_EQ(1u, audio_buffer->mNumberChannels);
+ RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
// Get pointer to internal audio buffer to which new audio data shall be
// written.
const size_t size_in_bytes = audio_buffer->mDataByteSize;
diff --git a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
index fca9a78..2550274 100644
--- a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
@@ -31,7 +31,7 @@
RTC_DCHECK_GE(target_gain, 0.0f);
size_t samples = audio_frame->samples_per_channel_;
- RTC_DCHECK_LT(0u, samples);
+ RTC_DCHECK_LT(0, samples);
float increment = (target_gain - start_gain) / samples;
float gain = start_gain;
for (size_t i = 0; i < samples; ++i) {
@@ -45,8 +45,8 @@
}
void RemixFrame(size_t target_number_of_channels, AudioFrame* frame) {
- RTC_DCHECK_GE(target_number_of_channels, 1u);
- RTC_DCHECK_LE(target_number_of_channels, 2u);
+ RTC_DCHECK_GE(target_number_of_channels, 1);
+ RTC_DCHECK_LE(target_number_of_channels, 2);
if (frame->num_channels_ == 1 && target_number_of_channels == 2) {
AudioFrameOperations::MonoToStereo(frame);
} else if (frame->num_channels_ == 2 && target_number_of_channels == 1) {
diff --git a/webrtc/modules/audio_processing/aec/aec_core.cc b/webrtc/modules/audio_processing/aec/aec_core.cc
index e3fd14c..b410c55 100644
--- a/webrtc/modules/audio_processing/aec/aec_core.cc
+++ b/webrtc/modules/audio_processing/aec/aec_core.cc
@@ -202,7 +202,7 @@
void BlockBuffer::ExtractExtendedBlock(float extended_block[PART_LEN2]) {
float* block_ptr = NULL;
- RTC_DCHECK_LT(0u, AvaliableSpace());
+ RTC_DCHECK_LT(0, AvaliableSpace());
// Extract the previous block.
WebRtc_MoveReadPtr(buffer_, -1);
@@ -461,7 +461,7 @@
// Average.
metric->counter++;
// This is to protect overflow, which should almost never happen.
- RTC_CHECK_NE(0u, metric->counter);
+ RTC_CHECK_NE(0, metric->counter);
metric->sum += metric->instant;
metric->average = metric->sum / metric->counter;
@@ -469,7 +469,7 @@
if (metric->instant > metric->average) {
metric->hicounter++;
// This is to protect overflow, which should almost never happen.
- RTC_CHECK_NE(0u, metric->hicounter);
+ RTC_CHECK_NE(0, metric->hicounter);
metric->hisum += metric->instant;
metric->himean = metric->hisum / metric->hicounter;
}
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.cc b/webrtc/modules/audio_processing/aec/aec_resampler.cc
index 2fde934..2630841 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.cc
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.cc
@@ -74,7 +74,7 @@
float be, tnew;
size_t tn, mm;
- RTC_DCHECK_LE(size, 2u * FRAME_LEN);
+ RTC_DCHECK_LE(size, 2 * FRAME_LEN);
RTC_DCHECK(resampInst);
RTC_DCHECK(inspeech);
RTC_DCHECK(outspeech);
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc b/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
index bc368f2..a34bcab 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
@@ -104,9 +104,9 @@
void WebRtcAecm_StoreAdaptiveChannelNeon(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est) {
- RTC_DCHECK_EQ(0u, (uintptr_t)echo_est % 32);
- RTC_DCHECK_EQ(0u, (uintptr_t)aecm->channelStored % 16);
- RTC_DCHECK_EQ(0u, (uintptr_t)aecm->channelAdapt16 % 16);
+ RTC_DCHECK_EQ(0, (uintptr_t)echo_est % 32);
+ RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
+ RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
// This is C code of following optimized code.
// During startup we store the channel every block.
@@ -161,9 +161,9 @@
}
void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore* aecm) {
- RTC_DCHECK_EQ(0u, (uintptr_t)aecm->channelStored % 16);
- RTC_DCHECK_EQ(0u, (uintptr_t)aecm->channelAdapt16 % 16);
- RTC_DCHECK_EQ(0u, (uintptr_t)aecm->channelAdapt32 % 32);
+ RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
+ RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
+ RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt32 % 32);
// The C code of following optimized code.
// for (i = 0; i < PART_LEN1; i++) {
diff --git a/webrtc/modules/audio_processing/agc/agc.cc b/webrtc/modules/audio_processing/agc/agc.cc
index a6256cb..3eca148 100644
--- a/webrtc/modules/audio_processing/agc/agc.cc
+++ b/webrtc/modules/audio_processing/agc/agc.cc
@@ -39,7 +39,7 @@
Agc::~Agc() {}
float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
- RTC_DCHECK_GT(length, 0u);
+ RTC_DCHECK_GT(length, 0);
size_t num_clipped = 0;
for (size_t i = 0; i < length; ++i) {
if (audio[i] == 32767 || audio[i] == -32768)
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index f5b9016..02b8537 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -62,11 +62,11 @@
activity_(AudioFrame::kVadUnknown),
keyboard_data_(NULL),
data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
- RTC_DCHECK_GT(input_num_frames_, 0u);
- RTC_DCHECK_GT(proc_num_frames_, 0u);
- RTC_DCHECK_GT(output_num_frames_, 0u);
- RTC_DCHECK_GT(num_input_channels_, 0u);
- RTC_DCHECK_GT(num_proc_channels_, 0u);
+ RTC_DCHECK_GT(input_num_frames_, 0);
+ RTC_DCHECK_GT(proc_num_frames_, 0);
+ RTC_DCHECK_GT(output_num_frames_, 0);
+ RTC_DCHECK_GT(num_input_channels_, 0);
+ RTC_DCHECK_GT(num_proc_channels_, 0);
RTC_DCHECK_LE(num_proc_channels_, num_input_channels_);
if (input_num_frames_ != proc_num_frames_ ||
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 39c79cd..061495d 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -813,7 +813,7 @@
num_reverse_channels(),
&aec_render_queue_buffer_);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
// Insert the samples into the queue.
if (!aec_render_signal_queue_->Insert(&aec_render_queue_buffer_)) {
diff --git a/webrtc/modules/audio_processing/beamformer/array_util.cc b/webrtc/modules/audio_processing/beamformer/array_util.cc
index 6b1c474..244f5c8 100644
--- a/webrtc/modules/audio_processing/beamformer/array_util.cc
+++ b/webrtc/modules/audio_processing/beamformer/array_util.cc
@@ -23,7 +23,7 @@
} // namespace
float GetMinimumSpacing(const std::vector<Point>& array_geometry) {
- RTC_CHECK_GT(array_geometry.size(), 1u);
+ RTC_CHECK_GT(array_geometry.size(), 1);
float mic_spacing = std::numeric_limits<float>::max();
for (size_t i = 0; i < (array_geometry.size() - 1); ++i) {
for (size_t j = i + 1; j < array_geometry.size(); ++j) {
@@ -58,7 +58,7 @@
rtc::Optional<Point> GetDirectionIfLinear(
const std::vector<Point>& array_geometry) {
- RTC_DCHECK_GT(array_geometry.size(), 1u);
+ RTC_DCHECK_GT(array_geometry.size(), 1);
const Point first_pair_direction =
PairDirection(array_geometry[0], array_geometry[1]);
for (size_t i = 2u; i < array_geometry.size(); ++i) {
@@ -73,7 +73,7 @@
rtc::Optional<Point> GetNormalIfPlanar(
const std::vector<Point>& array_geometry) {
- RTC_DCHECK_GT(array_geometry.size(), 1u);
+ RTC_DCHECK_GT(array_geometry.size(), 1);
const Point first_pair_direction =
PairDirection(array_geometry[0], array_geometry[1]);
Point pair_direction(0.f, 0.f, 0.f);
diff --git a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
index 5ca79c6..ae69073 100644
--- a/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
+++ b/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
@@ -27,7 +27,7 @@
// Calculates the Euclidean norm for a row vector.
float Norm(const ComplexMatrix<float>& x) {
- RTC_CHECK_EQ(1u, x.num_rows());
+ RTC_CHECK_EQ(1, x.num_rows());
const size_t length = x.num_columns();
const complex<float>* elems = x.elements()[0];
float result = 0.f;
@@ -94,7 +94,7 @@
const std::vector<Point>& geometry,
float angle,
ComplexMatrix<float>* mat) {
- RTC_CHECK_EQ(1u, mat->num_rows());
+ RTC_CHECK_EQ(1, mat->num_rows());
RTC_CHECK_EQ(geometry.size(), mat->num_columns());
float freq_in_hertz =
diff --git a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
index 5d9d32a..425ffa0 100644
--- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
+++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
@@ -79,7 +79,7 @@
// The returned norm is clamped to be non-negative.
float Norm(const ComplexMatrix<float>& mat,
const ComplexMatrix<float>& norm_mat) {
- RTC_CHECK_EQ(1u, norm_mat.num_rows());
+ RTC_CHECK_EQ(1, norm_mat.num_rows());
RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
@@ -102,8 +102,8 @@
// Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|.
complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
const ComplexMatrix<float>& rhs) {
- RTC_CHECK_EQ(1u, lhs.num_rows());
- RTC_CHECK_EQ(1u, rhs.num_rows());
+ RTC_CHECK_EQ(1, lhs.num_rows());
+ RTC_CHECK_EQ(1, rhs.num_rows());
RTC_CHECK_EQ(lhs.num_columns(), rhs.num_columns());
const complex<float>* const* lhs_elements = lhs.elements();
@@ -138,7 +138,7 @@
// Does |out| = |in|.' * conj(|in|) for row vector |in|.
void TransposedConjugatedProduct(const ComplexMatrix<float>& in,
ComplexMatrix<float>* out) {
- RTC_CHECK_EQ(1u, in.num_rows());
+ RTC_CHECK_EQ(1, in.num_rows());
RTC_CHECK_EQ(out->num_rows(), in.num_columns());
RTC_CHECK_EQ(out->num_columns(), in.num_columns());
const complex<float>* in_elements = in.elements()[0];
@@ -449,7 +449,7 @@
complex_f* const* output) {
RTC_CHECK_EQ(kNumFreqBins, num_freq_bins);
RTC_CHECK_EQ(num_input_channels_, num_input_channels);
- RTC_CHECK_EQ(0u, num_output_channels);
+ RTC_CHECK_EQ(0, num_output_channels);
// Calculating the post-filter masks. Note that we need two for each
// frequency bin to account for the positive and negative interferer
diff --git a/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
index 0b8fc14..f6a0bcd 100644
--- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc
+++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc
@@ -151,7 +151,7 @@
}
RTC_DCHECK(stream_properties_);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_proc_channels);
int err = AudioProcessing::kNoError;
@@ -450,7 +450,7 @@
size_t num_output_channels,
size_t num_channels,
std::vector<float>* packed_buffer) {
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(num_channels, audio->num_channels());
packed_buffer->clear();
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index cfc4249..e8b163b 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -154,7 +154,7 @@
size_t num_output_channels,
size_t num_channels,
std::vector<int16_t>* packed_buffer) {
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(num_channels, audio->num_channels());
// The ordering convention must be followed to pass to the correct AECM.
@@ -187,7 +187,7 @@
}
RTC_DCHECK(stream_properties_);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels);
RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels *
audio->num_channels());
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 81469dd..704cfad 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -123,7 +123,7 @@
void GainControlImpl::PackRenderAudioBuffer(
AudioBuffer* audio,
std::vector<int16_t>* packed_buffer) {
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
packed_buffer->clear();
packed_buffer->insert(
@@ -139,7 +139,7 @@
}
RTC_DCHECK(num_proc_channels_);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size());
@@ -190,7 +190,7 @@
}
RTC_DCHECK(num_proc_channels_);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
stream_is_saturated_ = false;
diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
index f9d1c3c..7ff2cf4 100644
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
@@ -58,7 +58,7 @@
const std::vector<std::vector<float>>& filter_bank,
float* result) {
for (size_t i = 0; i < filter_bank.size(); ++i) {
- RTC_DCHECK_GT(filter_bank[i].size(), 0u);
+ RTC_DCHECK_GT(filter_bank[i].size(), 0);
result[i] = kPowerNormalizationFactor *
DotProduct(filter_bank[i].data(), pow, filter_bank[i].size());
}
@@ -380,7 +380,7 @@
}
void IntelligibilityEnhancer::DelayHighBands(AudioBuffer* audio) {
- RTC_DCHECK_EQ(audio->num_bands(), high_bands_buffers_.size() + 1u);
+ RTC_DCHECK_EQ(audio->num_bands(), high_bands_buffers_.size() + 1);
for (size_t i = 0u; i < high_bands_buffers_.size(); ++i) {
Band band = static_cast<Band>(i + 1);
high_bands_buffers_[i]->Delay(audio->split_channels_f(band), chunk_length_);
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.cc b/webrtc/modules/audio_processing/level_controller/level_controller.cc
index b8388e6..c625e08 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.cc
@@ -208,8 +208,8 @@
}
void LevelController::Process(AudioBuffer* audio) {
- RTC_DCHECK_LT(0u, audio->num_channels());
- RTC_DCHECK_GE(2u, audio->num_channels());
+ RTC_DCHECK_LT(0, audio->num_channels());
+ RTC_DCHECK_GE(2, audio->num_channels());
RTC_DCHECK_NE(0.f, dc_forgetting_factor_);
RTC_DCHECK(sample_rate_hz_);
data_dumper_->DumpWav("lc_input", audio->num_frames(),
diff --git a/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc b/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
index af71868..2fbe200 100644
--- a/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
+++ b/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
@@ -34,7 +34,7 @@
void NoiseSpectrumEstimator::Update(rtc::ArrayView<const float> spectrum,
bool first_update) {
- RTC_DCHECK_EQ(65u, spectrum.size());
+ RTC_DCHECK_EQ(65, spectrum.size());
if (first_update) {
// Initialize the noise spectral estimate with the signal spectrum.
diff --git a/webrtc/modules/audio_processing/level_controller/signal_classifier.cc b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
index dd67737..f38dfb2 100644
--- a/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
+++ b/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
@@ -25,7 +25,7 @@
namespace {
void RemoveDcLevel(rtc::ArrayView<float> x) {
- RTC_DCHECK_LT(0u, x.size());
+ RTC_DCHECK_LT(0, x.size());
float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f);
mean /= x.size();
@@ -37,8 +37,8 @@
void PowerSpectrum(const OouraFft* ooura_fft,
rtc::ArrayView<const float> x,
rtc::ArrayView<float> spectrum) {
- RTC_DCHECK_EQ(65u, spectrum.size());
- RTC_DCHECK_EQ(128u, x.size());
+ RTC_DCHECK_EQ(65, spectrum.size());
+ RTC_DCHECK_EQ(128, x.size());
float X[128];
std::copy(x.data(), x.data() + x.size(), X);
ooura_fft->Fft(X);
diff --git a/webrtc/modules/audio_processing/low_cut_filter.cc b/webrtc/modules/audio_processing/low_cut_filter.cc
index 77dab9a..703535c 100644
--- a/webrtc/modules/audio_processing/low_cut_filter.cc
+++ b/webrtc/modules/audio_processing/low_cut_filter.cc
@@ -91,7 +91,7 @@
void LowCutFilter::Process(AudioBuffer* audio) {
RTC_DCHECK(audio);
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
for (size_t i = 0; i < filters_.size(); i++) {
filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index e1c9fdc..628b951 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -76,7 +76,7 @@
return;
}
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
WebRtcNs_Analyze(suppressors_[i]->state(),
@@ -92,7 +92,7 @@
return;
}
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
for (size_t i = 0; i < suppressors_.size(); i++) {
#if defined(WEBRTC_NS_FLOAT)
diff --git a/webrtc/modules/audio_processing/residual_echo_detector.cc b/webrtc/modules/audio_processing/residual_echo_detector.cc
index 0e58b49..45ef180 100644
--- a/webrtc/modules/audio_processing/residual_echo_detector.cc
+++ b/webrtc/modules/audio_processing/residual_echo_detector.cc
@@ -130,7 +130,7 @@
void ResidualEchoDetector::PackRenderAudioBuffer(
AudioBuffer* audio,
std::vector<float>* packed_buffer) {
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
packed_buffer->clear();
packed_buffer->insert(packed_buffer->end(),
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 95490f0..24e9b0e 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -136,7 +136,7 @@
const std::vector<float> values = ParseList<float>(mic_positions);
const size_t num_mics =
rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
- RTC_CHECK_GT(num_mics, 0u) << "mic_positions is not large enough.";
+ RTC_CHECK_GT(num_mics, 0) << "mic_positions is not large enough.";
std::vector<Point> result;
result.reserve(num_mics);
diff --git a/webrtc/modules/audio_processing/transient/moving_moments.cc b/webrtc/modules/audio_processing/transient/moving_moments.cc
index bc0b6f0..8bca505 100644
--- a/webrtc/modules/audio_processing/transient/moving_moments.cc
+++ b/webrtc/modules/audio_processing/transient/moving_moments.cc
@@ -22,7 +22,7 @@
queue_(),
sum_(0.0),
sum_of_squares_(0.0) {
- RTC_DCHECK_GT(length, 0u);
+ RTC_DCHECK_GT(length, 0);
for (size_t i = 0; i < length; ++i) {
queue_.push(0.0);
}
diff --git a/webrtc/modules/audio_processing/transient/wpd_node.cc b/webrtc/modules/audio_processing/transient/wpd_node.cc
index a689827..de382b4 100644
--- a/webrtc/modules/audio_processing/transient/wpd_node.cc
+++ b/webrtc/modules/audio_processing/transient/wpd_node.cc
@@ -29,9 +29,9 @@
filter_(FIRFilter::Create(coefficients,
coefficients_length,
2 * length + 1)) {
- RTC_DCHECK_GT(length, 0u);
+ RTC_DCHECK_GT(length, 0);
RTC_DCHECK(coefficients);
- RTC_DCHECK_GT(coefficients_length, 0u);
+ RTC_DCHECK_GT(coefficients_length, 0);
memset(data_.get(), 0.f, (2 * length + 1) * sizeof(data_[0]));
}
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.cc b/webrtc/modules/audio_processing/voice_detection_impl.cc
index a0702e8..5365ed0 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -63,7 +63,7 @@
return;
}
- RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+ RTC_DCHECK_GE(160, audio->num_frames_per_band());
// TODO(ajm): concatenate data in frame buffer here.
int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
audio->mixed_low_pass_data(),
diff --git a/webrtc/modules/pacing/bitrate_prober.cc b/webrtc/modules/pacing/bitrate_prober.cc
index fde2f65..431aa21 100644
--- a/webrtc/modules/pacing/bitrate_prober.cc
+++ b/webrtc/modules/pacing/bitrate_prober.cc
@@ -27,7 +27,7 @@
constexpr int kMinProbeDeltaMs = 1;
int ComputeDeltaFromBitrate(size_t probe_size, uint32_t bitrate_bps) {
- RTC_CHECK_GT(bitrate_bps, 0u);
+ RTC_CHECK_GT(bitrate_bps, 0);
// Compute the time delta needed to send probe_size bytes at bitrate_bps
// bps. Result is in milliseconds.
return static_cast<int>((1000ll * probe_size * 8) / bitrate_bps);
@@ -153,7 +153,7 @@
void BitrateProber::ProbeSent(int64_t now_ms, size_t bytes) {
RTC_DCHECK(probing_state_ == ProbingState::kActive);
- RTC_DCHECK_GT(bytes, 0u);
+ RTC_DCHECK_GT(bytes, 0);
probe_size_last_sent_ = bytes;
time_last_probe_sent_ms_ = now_ms;
if (!clusters_.empty()) {
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 00523e6..b3fc16d 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -129,7 +129,7 @@
packet_list_.erase(packet.this_it);
RTC_DCHECK_EQ(packet_list_.size(), prio_queue_.size());
if (packet_list_.empty())
- RTC_DCHECK_EQ(0u, queue_time_sum_);
+ RTC_DCHECK_EQ(0, queue_time_sum_);
}
bool Empty() const { return prio_queue_.empty(); }
@@ -285,7 +285,7 @@
}
void PacedSender::SetProbingEnabled(bool enabled) {
- RTC_CHECK_EQ(0u, packet_counter_);
+ RTC_CHECK_EQ(0, packet_counter_);
CriticalSectionScoped cs(critsect_.get());
prober_->SetEnabled(enabled);
}
@@ -338,7 +338,7 @@
int64_t PacedSender::ExpectedQueueTimeMs() const {
CriticalSectionScoped cs(critsect_.get());
- RTC_DCHECK_GT(pacing_bitrate_kbps_, 0u);
+ RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
return static_cast<int64_t>(packets_->SizeInBytes() * 8 /
pacing_bitrate_kbps_);
}
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
index 3c1a914..b77550f 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
@@ -247,7 +247,7 @@
// target bitrate after the call to this function.
bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(uint32_t ssrc,
uint32_t bitrate_bps) {
- RTC_DCHECK_GT(bitrate_bps, 0u);
+ RTC_DCHECK_GT(bitrate_bps, 0);
stream_generator_->SetBitrateBps(bitrate_bps);
testing::RtpStream::PacketList packets;
int64_t next_time_us = stream_generator_->GenerateFrame(
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index 568ba8a..7a3b158 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -103,7 +103,7 @@
const size_t num_media_packets = media_packets.size();
// Sanity check arguments.
- RTC_DCHECK_GT(num_media_packets, 0u);
+ RTC_DCHECK_GT(num_media_packets, 0);
RTC_DCHECK_GE(num_important_packets, 0);
RTC_DCHECK_LE(static_cast<size_t>(num_important_packets), num_media_packets);
RTC_DCHECK(fec_packets->empty());
@@ -239,7 +239,7 @@
pkt_mask_idx += media_pkt_idx / 8;
media_pkt_idx %= 8;
}
- RTC_DCHECK_GT(fec_packet->length, 0u)
+ RTC_DCHECK_GT(fec_packet->length, 0)
<< "Packet mask is wrong or poorly designed.";
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
index 5339c65..a2d99e7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
@@ -50,8 +50,8 @@
size_t RtcpPacket::HeaderLength() const {
size_t length_in_bytes = BlockLength();
- RTC_DCHECK_GT(length_in_bytes, 0u);
- RTC_DCHECK_EQ(length_in_bytes % 4, 0u) << "Padding not supported";
+ RTC_DCHECK_GT(length_in_bytes, 0);
+ RTC_DCHECK_EQ(length_in_bytes % 4, 0) << "Padding not supported";
// Length in 32-bit words without common header.
return (length_in_bytes - kHeaderLength) / 4;
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
index a2d37a4..322bf36 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
@@ -58,7 +58,7 @@
void App::SetData(const uint8_t* data, size_t data_length) {
RTC_DCHECK(data);
- RTC_DCHECK_EQ(data_length % 4, 0u) << "Data must be 32 bits aligned.";
+ RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
RTC_DCHECK_LE(data_length, kMaxDataSize) << "App data size " << data_length
<< " exceed maximum of "
<< kMaxDataSize << " bytes.";
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
index cdb45b0..494c870 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc
@@ -101,7 +101,7 @@
*index += reason_length;
// Add padding bytes if needed.
size_t bytes_to_pad = index_end - *index;
- RTC_DCHECK_LE(bytes_to_pad, 3u);
+ RTC_DCHECK_LE(bytes_to_pad, 3);
if (bytes_to_pad > 0) {
memset(&packet[*index], 0, bytes_to_pad);
*index += bytes_to_pad;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc
index 69df03a..7f62e18 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc
@@ -84,7 +84,7 @@
size_t index_end = *index + BlockLength();
CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
index);
- RTC_DCHECK_EQ(Psfb::media_ssrc(), 0u);
+ RTC_DCHECK_EQ(Psfb::media_ssrc(), 0);
CreateCommonFeedback(packet + *index);
*index += kCommonFeedbackLength;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
index ac5fc1a..ec7b51a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc
@@ -103,7 +103,7 @@
size_t index_end = *index + BlockLength();
CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
index);
- RTC_DCHECK_EQ(0u, Psfb::media_ssrc());
+ RTC_DCHECK_EQ(0, Psfb::media_ssrc());
CreateCommonFeedback(packet + *index);
*index += kCommonFeedbackLength;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
index 8015fa3..e597748 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
@@ -65,7 +65,7 @@
void ReportBlock::Create(uint8_t* buffer) const {
// Runtime check should be done while setting cumulative_lost.
- RTC_DCHECK_LT(cumulative_lost(), (1u << 24)); // Have only 3 bytes for it.
+ RTC_DCHECK_LT(cumulative_lost(), (1 << 24)); // Have only 3 bytes for it.
ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
index a0b785c..702afd8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc
@@ -86,7 +86,7 @@
CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
index);
- RTC_DCHECK_EQ(0u, Rtpfb::media_ssrc());
+ RTC_DCHECK_EQ(0, Rtpfb::media_ssrc());
CreateCommonFeedback(packet + *index);
*index += kCommonFeedbackLength;
for (const TmmbItem& item : items_) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
index b4b8ccb..0ba1311 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc
@@ -88,7 +88,7 @@
CreateHeader(kFeedbackMessageType, kPacketType, HeaderLength(), packet,
index);
- RTC_DCHECK_EQ(0u, Rtpfb::media_ssrc());
+ RTC_DCHECK_EQ(0, Rtpfb::media_ssrc());
CreateCommonFeedback(packet + *index);
*index += kCommonFeedbackLength;
for (const TmmbItem& item : items_) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
index 9008aa0..3a5a303 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
@@ -141,13 +141,13 @@
buffer[0] = 0x80u;
for (int i = 0; i < kSymbolsInFirstByte; ++i) {
uint8_t encoded_symbol = EncodeSymbol(symbols_[i]);
- RTC_DCHECK_LE(encoded_symbol, 1u);
+ RTC_DCHECK_LE(encoded_symbol, 1);
buffer[0] |= encoded_symbol << (kSymbolsInFirstByte - (i + 1));
}
buffer[1] = 0x00u;
for (int i = 0; i < kSymbolsInSecondByte; ++i) {
uint8_t encoded_symbol = EncodeSymbol(symbols_[i + kSymbolsInFirstByte]);
- RTC_DCHECK_LE(encoded_symbol, 1u);
+ RTC_DCHECK_LE(encoded_symbol, 1);
buffer[1] |= encoded_symbol << (kSymbolsInSecondByte - (i + 1));
}
}
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index b32e78e..2344b28 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -194,7 +194,7 @@
offset += packet_length;
fragment_length -= packet_length;
}
- RTC_CHECK_EQ(0u, fragment_length);
+ RTC_CHECK_EQ(0, fragment_length);
}
size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
@@ -205,7 +205,7 @@
const Fragment* fragment = &input_fragments_[fragment_index];
RTC_CHECK_GE(payload_size_left, fragment->length);
while (payload_size_left >= fragment->length + fragment_headers_length) {
- RTC_CHECK_GT(fragment->length, 0u);
+ RTC_CHECK_GT(fragment->length, 0);
packets_.push(PacketUnit(*fragment, aggregated_fragments == 0, false, true,
fragment->buffer[0]));
payload_size_left -= fragment->length;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
index 245ea8b..72fd789 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -253,9 +253,9 @@
}
void Packet::SetCsrcs(const std::vector<uint32_t>& csrcs) {
- RTC_DCHECK_EQ(num_extensions_, 0u);
- RTC_DCHECK_EQ(payload_size_, 0u);
- RTC_DCHECK_EQ(padding_size_, 0u);
+ RTC_DCHECK_EQ(num_extensions_, 0);
+ RTC_DCHECK_EQ(payload_size_, 0);
+ RTC_DCHECK_EQ(padding_size_, 0);
RTC_DCHECK_LE(csrcs.size(), 0x0fu);
RTC_DCHECK_LE(kFixedHeaderSize + 4 * csrcs.size(), capacity());
payload_offset_ = kFixedHeaderSize + 4 * csrcs.size();
@@ -269,7 +269,7 @@
}
uint8_t* Packet::AllocatePayload(size_t size_bytes) {
- RTC_DCHECK_EQ(padding_size_, 0u);
+ RTC_DCHECK_EQ(padding_size_, 0);
if (payload_offset_ + size_bytes > capacity()) {
LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
return nullptr;
@@ -283,7 +283,7 @@
}
void Packet::SetPayloadSize(size_t size_bytes) {
- RTC_DCHECK_EQ(padding_size_, 0u);
+ RTC_DCHECK_EQ(padding_size_, 0);
RTC_DCHECK_LE(size_bytes, payload_size_);
payload_size_ = size_bytes;
buffer_.SetSize(payload_offset_ + payload_size_);
@@ -473,8 +473,8 @@
if (extension_id == ExtensionManager::kInvalidId) {
return false;
}
- RTC_DCHECK_GT(length, 0u);
- RTC_DCHECK_LE(length, 16u);
+ RTC_DCHECK_GT(length, 0);
+ RTC_DCHECK_LE(length, 16);
size_t num_csrc = data()[0] & 0x0F;
size_t extensions_offset = kFixedHeaderSize + (num_csrc * 4) + 4;
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
index 0a15209..c4edc73 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
@@ -45,7 +45,7 @@
}
void RtpPacketHistory::Allocate(size_t number_to_store) {
- RTC_DCHECK_GT(number_to_store, 0u);
+ RTC_DCHECK_GT(number_to_store, 0);
RTC_DCHECK_LE(number_to_store, kMaxCapacity);
store_ = true;
stored_packets_.resize(number_to_store);
diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index 5842b90..8f83e05 100644
--- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -53,7 +53,7 @@
void InitFrame(AudioFrame* frame, size_t channels, size_t samples_per_channel,
int16_t left_data, int16_t right_data) {
RTC_DCHECK(frame);
- RTC_DCHECK_GE(2u, channels);
+ RTC_DCHECK_GE(2, channels);
RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples,
samples_per_channel * channels);
frame->samples_per_channel_ = samples_per_channel;
diff --git a/webrtc/modules/video_coding/codec_database.cc b/webrtc/modules/video_coding/codec_database.cc
index ff7077e..75e7043 100644
--- a/webrtc/modules/video_coding/codec_database.cc
+++ b/webrtc/modules/video_coding/codec_database.cc
@@ -198,7 +198,7 @@
RTC_DCHECK_GE(number_of_cores, 1);
RTC_DCHECK_GE(send_codec->plType, 1);
// Make sure the start bit rate is sane...
- RTC_DCHECK_LE(send_codec->startBitrate, 1000000u);
+ RTC_DCHECK_LE(send_codec->startBitrate, 1000000);
RTC_DCHECK(send_codec->codecType != kVideoCodecUnknown);
bool reset_required = pending_encoder_reset_;
if (number_of_cores_ != number_of_cores) {
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
index b8a2dcd..c5d2ed9 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
@@ -53,7 +53,7 @@
rtc::scoped_refptr<Vp9FrameBufferPool::Vp9FrameBuffer>
Vp9FrameBufferPool::GetFrameBuffer(size_t min_size) {
- RTC_DCHECK_GT(min_size, 0u);
+ RTC_DCHECK_GT(min_size, 0);
rtc::scoped_refptr<Vp9FrameBuffer> available_buffer = nullptr;
{
rtc::CritScope cs(&buffers_lock_);
diff --git a/webrtc/modules/video_coding/histogram.cc b/webrtc/modules/video_coding/histogram.cc
index f2aa6ea..f862faf 100644
--- a/webrtc/modules/video_coding/histogram.cc
+++ b/webrtc/modules/video_coding/histogram.cc
@@ -17,8 +17,8 @@
namespace webrtc {
namespace video_coding {
Histogram::Histogram(size_t num_buckets, size_t max_num_values) {
- RTC_DCHECK_GT(num_buckets, 0u);
- RTC_DCHECK_GT(max_num_values, 0u);
+ RTC_DCHECK_GT(num_buckets, 0);
+ RTC_DCHECK_GT(max_num_values, 0);
buckets_.resize(num_buckets);
values_.reserve(max_num_values);
index_ = 0;
diff --git a/webrtc/modules/video_coding/video_codec_initializer.cc b/webrtc/modules/video_coding/video_codec_initializer.cc
index 57f9b26..c6db916 100644
--- a/webrtc/modules/video_coding/video_codec_initializer.cc
+++ b/webrtc/modules/video_coding/video_codec_initializer.cc
@@ -177,8 +177,8 @@
}
for (size_t i = 0; i < streams.size(); ++i) {
SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
- RTC_DCHECK_GT(streams[i].width, 0u);
- RTC_DCHECK_GT(streams[i].height, 0u);
+ RTC_DCHECK_GT(streams[i].width, 0);
+ RTC_DCHECK_GT(streams[i].height, 0);
RTC_DCHECK_GT(streams[i].max_framerate, 0);
// Different framerates not supported per stream at the moment.
RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate);
diff --git a/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc b/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc
index 0d79758..d48e990 100644
--- a/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc
+++ b/webrtc/sdk/objc/Framework/Classes/h264_video_toolbox_nalu.cc
@@ -56,7 +56,7 @@
return false;
}
RTC_CHECK_EQ(nalu_header_size, kAvccHeaderByteSize);
- RTC_DCHECK_EQ(param_set_count, 2u);
+ RTC_DCHECK_EQ(param_set_count, 2);
// Truncate any previous data in the buffer without changing its capacity.
annexb_buffer->SetSize(0);
@@ -250,7 +250,7 @@
bool H264AnnexBBufferHasVideoFormatDescription(const uint8_t* annexb_buffer,
size_t annexb_buffer_size) {
RTC_DCHECK(annexb_buffer);
- RTC_DCHECK_GT(annexb_buffer_size, 4u);
+ RTC_DCHECK_GT(annexb_buffer_size, 4);
// The buffer we receive via RTP has 00 00 00 01 start code artifically
// embedded by the RTP depacketizer. Extract NALU information.
diff --git a/webrtc/system_wrappers/include/aligned_array.h b/webrtc/system_wrappers/include/aligned_array.h
index a2ffe99..71fefea 100644
--- a/webrtc/system_wrappers/include/aligned_array.h
+++ b/webrtc/system_wrappers/include/aligned_array.h
@@ -23,7 +23,7 @@
AlignedArray(size_t rows, size_t cols, size_t alignment)
: rows_(rows),
cols_(cols) {
- RTC_CHECK_GT(alignment, 0u);
+ RTC_CHECK_GT(alignment, 0);
head_row_ = static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_),
alignment));
for (size_t i = 0; i < rows_; ++i) {
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index aaafdb3..aadcb8d 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -203,8 +203,8 @@
size_t num_flexfec_streams,
Transport* send_transport) {
RTC_DCHECK(num_video_streams <= kNumSsrcs);
- RTC_DCHECK_LE(num_audio_streams, 1u);
- RTC_DCHECK_LE(num_flexfec_streams, 1u);
+ RTC_DCHECK_LE(num_audio_streams, 1);
+ RTC_DCHECK_LE(num_flexfec_streams, 1);
RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
if (num_video_streams > 0) {
video_send_config_ = VideoSendStream::Config(send_transport);
@@ -261,7 +261,7 @@
}
}
- RTC_DCHECK_GE(1u, num_audio_streams_);
+ RTC_DCHECK_GE(1, num_audio_streams_);
if (num_audio_streams_ == 1) {
RTC_DCHECK_LE(0, voe_send_.channel_id);
AudioReceiveStream::Config audio_config;
diff --git a/webrtc/test/frame_generator.cc b/webrtc/test/frame_generator.cc
index eeb0b0f..b6307f9 100644
--- a/webrtc/test/frame_generator.cc
+++ b/webrtc/test/frame_generator.cc
@@ -161,7 +161,7 @@
current_source_frame_(nullptr),
file_generator_(files, source_width, source_height, 1) {
RTC_DCHECK(clock_ != nullptr);
- RTC_DCHECK_GT(num_frames_, 0u);
+ RTC_DCHECK_GT(num_frames_, 0);
RTC_DCHECK_GE(source_height, target_height);
RTC_DCHECK_GE(source_width, target_width);
RTC_DCHECK_GE(scroll_time_ms, 0);
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index cf8d5d0..86c181b 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1039,7 +1039,7 @@
}
// Configure encoding and decoding with VP8, since generic packetization
// doesn't support FEC with NACK.
- RTC_DCHECK_EQ(1u, (*receive_configs)[0].decoders.size());
+ RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
(*receive_configs)[0].decoders[0].payload_name = "VP8";
@@ -2657,7 +2657,7 @@
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
- RTC_DCHECK_EQ(1u, encoder_config->number_of_streams);
+ RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
}
int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
@@ -3238,7 +3238,7 @@
if (encoder_config.number_of_streams > 1) {
// Lower bitrates so that all streams send initially.
- RTC_DCHECK_EQ(3u, encoder_config.number_of_streams);
+ RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index f7a63ae..8f5cb36 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -157,7 +157,7 @@
// spare cores.
uint32_t num_cores = CpuInfo::DetectNumberOfCores();
- RTC_DCHECK_GE(num_cores, 1u);
+ RTC_DCHECK_GE(num_cores, 1);
static const uint32_t kMinCoresLeft = 4;
static const uint32_t kMaxComparisonThreads = 8;
@@ -757,7 +757,7 @@
void AddCapturedFrameForComparison(const VideoFrame& video_frame) {
rtc::CritScope lock(&crit_);
- RTC_DCHECK_EQ(0u, video_frame.timestamp());
+ RTC_DCHECK_EQ(0, video_frame.timestamp());
// Frames from the capturer does not have a rtp timestamp. Create one so it
// can be used for comparison.
VideoFrame copy = video_frame;
@@ -891,7 +891,7 @@
if (params_.video.codec == "VP8") {
RTC_CHECK_EQ(params_.ss.num_spatial_layers, 1);
} else if (params_.video.codec == "VP9") {
- RTC_CHECK_EQ(params_.ss.streams.size(), 1u);
+ RTC_CHECK_EQ(params_.ss.streams.size(), 1);
}
}
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 331f009..0821059 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -50,7 +50,7 @@
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
size_t num_modules) {
- RTC_DCHECK_GT(num_modules, 0u);
+ RTC_DCHECK_GT(num_modules, 0);
RtpRtcp::Configuration configuration;
ReceiveStatistics* null_receive_statistics = configuration.receive_statistics;
configuration.audio = false;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 4412f4e..359a013 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -1110,7 +1110,7 @@
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
- RTC_DCHECK_EQ(1u, encoder_config->number_of_streams);
+ RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
transport_adapter_.reset(
new internal::TransportAdapter(send_config->send_transport));
transport_adapter_->Enable();
@@ -1554,7 +1554,7 @@
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
- RTC_DCHECK_EQ(1u, encoder_config->number_of_streams);
+ RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
if (running_without_padding_) {
encoder_config->min_transmit_bitrate_bps = 0;
encoder_config->content_type =
diff --git a/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc b/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
index 1787915..09ca127 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/loudest_filter.cc
@@ -67,7 +67,7 @@
}
unsigned int quietest_ssrc = FindQuietestStream();
- RTC_CHECK_NE(0u, quietest_ssrc);
+ RTC_CHECK_NE(0, quietest_ssrc);
// A smaller value if audio level corresponds to a louder sound.
if (audio_level < stream_levels_[quietest_ssrc].audio_level) {
stream_levels_.erase(quietest_ssrc);
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index 37e12ce..88c60fd 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -82,10 +82,10 @@
const int16_t source[],
size_t source_channel,
size_t source_len) {
- RTC_DCHECK_GE(target_channel, 1u);
- RTC_DCHECK_LE(target_channel, 2u);
- RTC_DCHECK_GE(source_channel, 1u);
- RTC_DCHECK_LE(source_channel, 2u);
+ RTC_DCHECK_GE(target_channel, 1);
+ RTC_DCHECK_LE(target_channel, 2);
+ RTC_DCHECK_GE(source_channel, 1);
+ RTC_DCHECK_LE(source_channel, 2);
if (target_channel == 2 && source_channel == 1) {
// Convert source from mono to stereo.
diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc
index bac9dca..fecb16a 100644
--- a/webrtc/voice_engine/voe_base_impl.cc
+++ b/webrtc/voice_engine/voe_base_impl.cc
@@ -535,7 +535,7 @@
}
std::string versionString = VoiceEngine::GetVersionString();
- RTC_DCHECK_GT(1024u, versionString.size() + 1);
+ RTC_DCHECK_GT(1024, versionString.size() + 1);
char* end = std::copy(versionString.cbegin(), versionString.cend(), version);
end[0] = '\n';
end[1] = '\0';