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RTC_[D]CHECK_op: Remove "u" suffix on integer constants

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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}
This commit is contained in:
kwiberg
2016-11-28 15:21:39 -08:00
committed by Commit bot
parent 80ed35e21c
commit af476c737f
89 changed files with 167 additions and 167 deletions
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2
webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
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@ -66,7 +66,7 @@ uint64_t DelayPeakDetector::MaxPeakPeriod() const {
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;
}

2
webrtc/modules/audio_coding/neteq/nack_tracker.cc
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@ -196,7 +196,7 @@ void NackTracker::Reset() {
}
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);

2
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
View File

@ -1724,7 +1724,7 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
#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;

2
webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
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@ -59,7 +59,7 @@ void EncodeNetEqInput::CreatePacket() {
// 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;

8
webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
View File

@ -26,13 +26,13 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
// 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 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
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 @@ void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
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));

2
webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
View File

@ -70,7 +70,7 @@ void NetEqReplacementInput::ReplacePacket() {
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.";