From af476c737fe008a185a99144731fd398e311ba65 Mon Sep 17 00:00:00 2001 From: kwiberg Date: Mon, 28 Nov 2016 15:21:39 -0800 Subject: [PATCH] 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} --- webrtc/audio/audio_transport_proxy.cc | 8 ++++---- webrtc/base/bitbuffer.cc | 4 ++-- webrtc/base/filerotatingstream.cc | 8 ++++---- webrtc/base/flags.cc | 2 +- webrtc/call/bitrate_allocator.cc | 4 ++-- webrtc/call/bitrate_estimator_tests.cc | 2 +- webrtc/call/call_perf_tests.cc | 2 +- webrtc/common_audio/audio_converter.cc | 2 +- webrtc/common_audio/channel_buffer.h | 2 +- webrtc/common_audio/include/audio_util.h | 2 +- webrtc/common_audio/lapped_transform.cc | 6 +++--- webrtc/common_audio/resampler/push_resampler.cc | 4 ++-- webrtc/common_audio/sparse_fir_filter.cc | 4 ++-- webrtc/common_audio/wav_file.cc | 4 ++-- webrtc/media/engine/webrtcvideoengine2.cc | 4 ++-- .../audio_coding/acm2/audio_coding_module.cc | 2 +- webrtc/modules/audio_coding/acm2/codec_manager.cc | 2 +- .../audio_coding/codecs/cng/audio_encoder_cng.cc | 4 ++-- .../audio_coding/codecs/g711/audio_decoder_pcm.h | 4 ++-- .../audio_coding/codecs/ilbc/audio_decoder_ilbc.cc | 2 +- .../audio_coding/codecs/opus/audio_encoder_opus.cc | 2 +- .../codecs/opus/audio_encoder_opus_unittest.cc | 2 +- .../codecs/pcm16b/audio_decoder_pcm16b.cc | 2 +- .../codecs/red/audio_encoder_copy_red.cc | 4 ++-- .../audio_coding/neteq/delay_peak_detector.cc | 2 +- webrtc/modules/audio_coding/neteq/nack_tracker.cc | 2 +- .../modules/audio_coding/neteq/test/RTPencode.cc | 2 +- .../audio_coding/neteq/tools/encode_neteq_input.cc | 2 +- .../neteq/tools/fake_decode_from_file.cc | 8 ++++---- .../neteq/tools/neteq_replacement_input.cc | 2 +- .../audio_device/android/opensles_player.cc | 4 ++-- webrtc/modules/audio_device/audio_device_buffer.cc | 2 +- .../modules/audio_device/ios/audio_device_ios.mm | 4 ++-- .../modules/audio_mixer/audio_frame_manipulator.cc | 6 +++--- webrtc/modules/audio_processing/aec/aec_core.cc | 6 +++--- .../modules/audio_processing/aec/aec_resampler.cc | 2 +- .../audio_processing/aecm/aecm_core_neon.cc | 12 ++++++------ webrtc/modules/audio_processing/agc/agc.cc | 2 +- webrtc/modules/audio_processing/audio_buffer.cc | 10 +++++----- .../audio_processing/audio_processing_impl.cc | 2 +- .../audio_processing/beamformer/array_util.cc | 6 +++--- .../beamformer/covariance_matrix_generator.cc | 4 ++-- .../beamformer/nonlinear_beamformer.cc | 10 +++++----- .../audio_processing/echo_cancellation_impl.cc | 4 ++-- .../audio_processing/echo_control_mobile_impl.cc | 4 ++-- .../modules/audio_processing/gain_control_impl.cc | 6 +++--- .../intelligibility/intelligibility_enhancer.cc | 4 ++-- .../level_controller/level_controller.cc | 4 ++-- .../level_controller/noise_spectrum_estimator.cc | 2 +- .../level_controller/signal_classifier.cc | 6 +++--- webrtc/modules/audio_processing/low_cut_filter.cc | 2 +- .../audio_processing/noise_suppression_impl.cc | 4 ++-- .../audio_processing/residual_echo_detector.cc | 2 +- webrtc/modules/audio_processing/test/test_utils.cc | 2 +- .../audio_processing/transient/moving_moments.cc | 2 +- .../modules/audio_processing/transient/wpd_node.cc | 4 ++-- .../audio_processing/voice_detection_impl.cc | 2 +- webrtc/modules/pacing/bitrate_prober.cc | 4 ++-- webrtc/modules/pacing/paced_sender.cc | 6 +++--- .../remote_bitrate_estimator_unittest_helper.cc | 2 +- .../rtp_rtcp/source/forward_error_correction.cc | 4 ++-- webrtc/modules/rtp_rtcp/source/rtcp_packet.cc | 4 ++-- webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc | 2 +- webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.cc | 2 +- webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.cc | 2 +- webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.cc | 2 +- .../rtp_rtcp/source/rtcp_packet/report_block.cc | 2 +- .../modules/rtp_rtcp/source/rtcp_packet/tmmbn.cc | 2 +- .../modules/rtp_rtcp/source/rtcp_packet/tmmbr.cc | 2 +- .../source/rtcp_packet/transport_feedback.cc | 4 ++-- webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc | 4 ++-- webrtc/modules/rtp_rtcp/source/rtp_packet.cc | 14 +++++++------- .../modules/rtp_rtcp/source/rtp_packet_history.cc | 2 +- .../source/audio_frame_operations_unittest.cc | 2 +- webrtc/modules/video_coding/codec_database.cc | 2 +- .../codecs/vp9/vp9_frame_buffer_pool.cc | 2 +- webrtc/modules/video_coding/histogram.cc | 4 ++-- .../video_coding/video_codec_initializer.cc | 4 ++-- .../Framework/Classes/h264_video_toolbox_nalu.cc | 4 ++-- webrtc/system_wrappers/include/aligned_array.h | 2 +- webrtc/test/call_test.cc | 6 +++--- webrtc/test/frame_generator.cc | 2 +- webrtc/video/end_to_end_tests.cc | 6 +++--- webrtc/video/video_quality_test.cc | 6 +++--- webrtc/video/video_send_stream.cc | 2 +- webrtc/video/video_send_stream_tests.cc | 4 ++-- .../test/auto_test/fakes/loudest_filter.cc | 2 +- webrtc/voice_engine/utility.cc | 8 ++++---- webrtc/voice_engine/voe_base_impl.cc | 2 +- 89 files changed, 167 insertions(+), 167 deletions(-) diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc index 163cc11c77..7036c1059c 100644 --- a/webrtc/audio/audio_transport_proxy.cc +++ b/webrtc/audio/audio_transport_proxy.cc @@ -68,8 +68,8 @@ int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples, 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(AudioProcessing::NativeRate::kSampleRate8kHz)); @@ -111,8 +111,8 @@ void AudioTransportProxy::PullRenderData(int bits_per_sample, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) { RTC_DCHECK_EQ(static_cast(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(sample_rate), AudioProcessing::NativeRate::kSampleRate8kHz); diff --git a/webrtc/base/bitbuffer.cc b/webrtc/base/bitbuffer.cc index 48a1d0ccdb..fc8a899ad6 100644 --- a/webrtc/base/bitbuffer.cc +++ b/webrtc/base/bitbuffer.cc @@ -19,14 +19,14 @@ namespace { // 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(bit_count); uint8_t mask = 0xFF << shift; return (byte & mask) >> shift; diff --git a/webrtc/base/filerotatingstream.cc b/webrtc/base/filerotatingstream.cc index 080999476b..d1434de9dd 100644 --- a/webrtc/base/filerotatingstream.cc +++ b/webrtc/base/filerotatingstream.cc @@ -37,8 +37,8 @@ FileRotatingStream::FileRotatingStream(const std::string& dir_path, 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 @@ bool FileRotatingStream::OpenCurrentFile() { 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 @@ CallSessionFileRotatingStream::CallSessionFileRotatingStream( 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 a138b8fb9b..4fcd4acc11 100644 --- a/webrtc/base/flags.cc +++ b/webrtc/base/flags.cc @@ -258,7 +258,7 @@ int FlagList::SetFlagsFromCommandLine(int* argc, const char** argv, 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 2c5ba6d76c..6e9be731d5 100644 --- a/webrtc/call/bitrate_allocator.cc +++ b/webrtc/call/bitrate_allocator.cc @@ -36,7 +36,7 @@ const int64_t kBweLogIntervalMs = 5000; 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 @@ void BitrateAllocator::DistributeBitrateEvenly(uint32_t bitrate, } 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(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 ddb69ed6fc..95d42ef117 100644 --- a/webrtc/call/bitrate_estimator_tests.cc +++ b/webrtc/call/bitrate_estimator_tests.cc @@ -166,7 +166,7 @@ class BitrateEstimatorTest : public test::CallTest { 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 354e092a4f..fb5ae0d45e 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -549,7 +549,7 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { 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 d1dcacd96d..231ba88077 100644 --- a/webrtc/common_audio/audio_converter.cc +++ b/webrtc/common_audio/audio_converter.cc @@ -109,7 +109,7 @@ class CompositionConverter : public AudioConverter { public: CompositionConverter(std::vector> 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 f4ed683484..c4c85d1ef5 100644 --- a/webrtc/common_audio/channel_buffer.h +++ b/webrtc/common_audio/channel_buffer.h @@ -94,7 +94,7 @@ class ChannelBuffer { // 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 e5ad701595..1601c7fd1e 100644 --- a/webrtc/common_audio/include/audio_util.h +++ b/webrtc/common_audio/include/audio_util.h @@ -154,7 +154,7 @@ void DownmixInterleavedToMonoImpl(const T* interleaved, 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 8a791f3291..6825bebb7a 100644 --- a/webrtc/common_audio/lapped_transform.cc +++ b/webrtc/common_audio/lapped_transform.cc @@ -84,12 +84,12 @@ LappedTransform::LappedTransform(size_t num_in_channels, 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 9f329c4cb9..788223d343 100644 --- a/webrtc/common_audio/resampler/push_resampler.cc +++ b/webrtc/common_audio/resampler/push_resampler.cc @@ -32,8 +32,8 @@ void CheckValidInitParams(int src_sample_rate_hz, int dst_sample_rate_hz, #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 a79da07b2d..2928004a0d 100644 --- a/webrtc/common_audio/sparse_fir_filter.cc +++ b/webrtc/common_audio/sparse_fir_filter.cc @@ -22,8 +22,8 @@ SparseFIRFilter::SparseFIRFilter(const float* nonzero_coeffs, 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 572c94ba3d..2b9098a6cd 100644 --- a/webrtc/common_audio/wav_file.cc +++ b/webrtc/common_audio/wav_file.cc @@ -123,7 +123,7 @@ WavWriter::WavWriter(const std::string& filename, int sample_rate, // 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 @@ void WavWriter::Close() { 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 e947fd1254..3f9d5f6596 100644 --- a/webrtc/media/engine/webrtcvideoengine2.cc +++ b/webrtc/media/engine/webrtcvideoengine2.cc @@ -1780,7 +1780,7 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec( 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 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::ReconfigureEncoder() { 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 ee1034b75a..dd02964e90 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 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) { 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 d8dcd792c4..afeefc78c2 100644 --- a/webrtc/modules/audio_coding/acm2/codec_manager.cc +++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc @@ -211,7 +211,7 @@ bool CodecManager::MakeEncoder(RentACodec* rac, AudioCodingModule* acm) { 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 d2edcb5c26..b1629410fc 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 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive( 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 483311b61b..b1d259e965 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 @@ namespace webrtc { 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 ParsePayload(rtc::Buffer&& payload, @@ -44,7 +44,7 @@ class AudioDecoderPcmU final : public AudioDecoder { 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 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 354f8194d8..a0fc02bb59 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 @@ std::vector AudioDecoderIlbc::ParsePayload( 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 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 e9772f674b..1dc312f5e9 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::SetFrameLength(int frame_length_ms) { } 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 54529ad985..8e59f496d8 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 @@ namespace { // Returns a vector with the n evenly-spaced numbers a, a + (b - a)/(n - 1), // ..., b. std::vector 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 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 43d2dacf9b..692e212ed9 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 @@ AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels) 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 37fa55a4da..07db78c911 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 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl( // 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 10535e23f5..5763572bad 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 @@ 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; } diff --git a/webrtc/modules/audio_coding/neteq/nack_tracker.cc b/webrtc/modules/audio_coding/neteq/nack_tracker.cc index 62cb2ec958..f97879319a 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::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); diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc index 4ccf4fbdcb..f390f5330b 100644 --- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc +++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc @@ -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; 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 c89200be87..263f7b4223 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 @@ 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; 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 29beed5644..2e452e1dea 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 @@ 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::ReadLittleEndian(encoded); uint32_t samples_to_decode = @@ -66,7 +66,7 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded, ByteReader::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 encoded) { - RTC_CHECK_GE(encoded.size(), 12u); + RTC_CHECK_GE(encoded.size(), 12); ByteWriter::WriteLittleEndian(&encoded[0], timestamp); ByteWriter::WriteLittleEndian(&encoded[4], rtc::checked_cast(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 c9163a15d8..dd10649625 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 @@ 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(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 d675d637e3..7dfc5ec891 100644 --- a/webrtc/modules/audio_device/android/opensles_player.cc +++ b/webrtc/modules/audio_device/android/opensles_player.cc @@ -145,8 +145,8 @@ int OpenSLESPlayer::StopPlayout() { // 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 83109177c6..e7a5da731a 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::RequestPlayoutData(size_t samples_per_channel) { 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 ed8dbad784..bf6cac7d3c 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 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags, 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 fca9a78b50..255027478a 100644 --- a/webrtc/modules/audio_mixer/audio_frame_manipulator.cc +++ b/webrtc/modules/audio_mixer/audio_frame_manipulator.cc @@ -31,7 +31,7 @@ void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { 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 Ramp(float start_gain, float target_gain, AudioFrame* audio_frame) { } 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 e3fd14c9da..b410c55507 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::Insert(const float block[PART_LEN]) { 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 @@ static void UpdateLogRatioMetric(Stats* metric, float numerator, // 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 @@ static void UpdateLogRatioMetric(Stats* metric, float numerator, 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 2fde934d99..2630841d4a 100644 --- a/webrtc/modules/audio_processing/aec/aec_resampler.cc +++ b/webrtc/modules/audio_processing/aec/aec_resampler.cc @@ -74,7 +74,7 @@ void WebRtcAec_ResampleLinear(void* resampInst, 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 bc368f207c..a34bcabfa2 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_CalcLinearEnergiesNeon(AecmCore* aecm, 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_StoreAdaptiveChannelNeon(AecmCore* aecm, } 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 a6256cbe46..3eca1483d5 100644 --- a/webrtc/modules/audio_processing/agc/agc.cc +++ b/webrtc/modules/audio_processing/agc/agc.cc @@ -39,7 +39,7 @@ Agc::Agc() 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 f5b9016a67..02b8537c07 100644 --- a/webrtc/modules/audio_processing/audio_buffer.cc +++ b/webrtc/modules/audio_processing/audio_buffer.cc @@ -62,11 +62,11 @@ AudioBuffer::AudioBuffer(size_t input_num_frames, 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 39c79cdf5c..061495d9a2 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -813,7 +813,7 @@ void AudioProcessingImpl::QueueRenderAudio(AudioBuffer* audio) { 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 6b1c474269..244f5c88e8 100644 --- a/webrtc/modules/audio_processing/beamformer/array_util.cc +++ b/webrtc/modules/audio_processing/beamformer/array_util.cc @@ -23,7 +23,7 @@ const float kMaxDotProduct = 1e-6f; } // namespace float GetMinimumSpacing(const std::vector& array_geometry) { - RTC_CHECK_GT(array_geometry.size(), 1u); + RTC_CHECK_GT(array_geometry.size(), 1); float mic_spacing = std::numeric_limits::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 @@ bool ArePerpendicular(const Point& a, const Point& b) { rtc::Optional GetDirectionIfLinear( const std::vector& 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 GetDirectionIfLinear( rtc::Optional GetNormalIfPlanar( const std::vector& 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 5ca79c6475..ae69073a89 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 @@ float BesselJ0(float x) { // Calculates the Euclidean norm for a row vector. float Norm(const ComplexMatrix& x) { - RTC_CHECK_EQ(1u, x.num_rows()); + RTC_CHECK_EQ(1, x.num_rows()); const size_t length = x.num_columns(); const complex* elems = x.elements()[0]; float result = 0.f; @@ -94,7 +94,7 @@ void CovarianceMatrixGenerator::PhaseAlignmentMasks( const std::vector& geometry, float angle, ComplexMatrix* 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 5d9d32a80d..425ffa0e0b 100644 --- a/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc +++ b/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc @@ -79,7 +79,7 @@ const float kCompensationGain = 2.f; // The returned norm is clamped to be non-negative. float Norm(const ComplexMatrix& mat, const ComplexMatrix& 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 @@ float Norm(const ComplexMatrix& mat, // Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|. complex ConjugateDotProduct(const ComplexMatrix& lhs, const ComplexMatrix& 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* const* lhs_elements = lhs.elements(); @@ -138,7 +138,7 @@ float SumSquares(const ComplexMatrix& mat) { // Does |out| = |in|.' * conj(|in|) for row vector |in|. void TransposedConjugatedProduct(const ComplexMatrix& in, ComplexMatrix* 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* in_elements = in.elements()[0]; @@ -449,7 +449,7 @@ void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input, 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 0b8fc141ee..f6a0bcde7a 100644 --- a/webrtc/modules/audio_processing/echo_cancellation_impl.cc +++ b/webrtc/modules/audio_processing/echo_cancellation_impl.cc @@ -151,7 +151,7 @@ int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio, } 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 @@ void EchoCancellationImpl::PackRenderAudioBuffer( size_t num_output_channels, size_t num_channels, std::vector* 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 cfc4249804..e8b163b894 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 @@ void EchoControlMobileImpl::PackRenderAudioBuffer( size_t num_output_channels, size_t num_channels, std::vector* 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 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio, } 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 81469dde01..704cfade09 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::ProcessRenderAudio( void GainControlImpl::PackRenderAudioBuffer( AudioBuffer* audio, std::vector* 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 @@ int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { } 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 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio, } 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 f9d1c3c273..7ff2cf4042 100644 --- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc +++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc @@ -58,7 +58,7 @@ void MapToErbBands(const float* pow, const std::vector>& 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 @@ bool IntelligibilityEnhancer::IsSpeech(const float* audio) { } 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(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 b8388e6141..c625e08eae 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::Initialize(int sample_rate_hz) { } 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 af718685cf..2fbe200fa7 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::Initialize() { void NoiseSpectrumEstimator::Update(rtc::ArrayView 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 dd67737403..f38dfb2e13 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 webrtc { namespace { void RemoveDcLevel(rtc::ArrayView 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 RemoveDcLevel(rtc::ArrayView x) { void PowerSpectrum(const OouraFft* ooura_fft, rtc::ArrayView x, rtc::ArrayView 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 77dab9af43..703535cb1d 100644 --- a/webrtc/modules/audio_processing/low_cut_filter.cc +++ b/webrtc/modules/audio_processing/low_cut_filter.cc @@ -91,7 +91,7 @@ LowCutFilter::~LowCutFilter() {} 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 e1c9fdcd01..628b951f57 100644 --- a/webrtc/modules/audio_processing/noise_suppression_impl.cc +++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc @@ -76,7 +76,7 @@ void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { 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 @@ void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { 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 0e58b497fe..45ef1809f0 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::Initialize() { void ResidualEchoDetector::PackRenderAudioBuffer( AudioBuffer* audio, std::vector* 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 95490f00a9..24e9b0ebd2 100644 --- a/webrtc/modules/audio_processing/test/test_utils.cc +++ b/webrtc/modules/audio_processing/test/test_utils.cc @@ -136,7 +136,7 @@ std::vector ParseArrayGeometry(const std::string& mic_positions) { const std::vector values = ParseList(mic_positions); const size_t num_mics = rtc::CheckedDivExact(values.size(), static_cast(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 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 bc0b6f0b91..8bca505aef 100644 --- a/webrtc/modules/audio_processing/transient/moving_moments.cc +++ b/webrtc/modules/audio_processing/transient/moving_moments.cc @@ -22,7 +22,7 @@ MovingMoments::MovingMoments(size_t length) 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 a689827e9a..de382b4242 100644 --- a/webrtc/modules/audio_processing/transient/wpd_node.cc +++ b/webrtc/modules/audio_processing/transient/wpd_node.cc @@ -29,9 +29,9 @@ WPDNode::WPDNode(size_t length, 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 a0702e868e..5365ed083c 100644 --- a/webrtc/modules/audio_processing/voice_detection_impl.cc +++ b/webrtc/modules/audio_processing/voice_detection_impl.cc @@ -63,7 +63,7 @@ void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { 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 fde2f65552..431aa21881 100644 --- a/webrtc/modules/pacing/bitrate_prober.cc +++ b/webrtc/modules/pacing/bitrate_prober.cc @@ -27,7 +27,7 @@ constexpr int kInactivityThresholdMs = 5000; 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((1000ll * probe_size * 8) / bitrate_bps); @@ -153,7 +153,7 @@ size_t BitrateProber::RecommendedMinProbeSize() const { 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 00523e61b7..b3fc16d769 100644 --- a/webrtc/modules/pacing/paced_sender.cc +++ b/webrtc/modules/pacing/paced_sender.cc @@ -129,7 +129,7 @@ class PacketQueue { 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::Resume() { } 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 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority, 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(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 3c1a914d28..b77550f121 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 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc, // 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 568ba8afbc..7a3b1583d8 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 @@ int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets, 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(num_important_packets), num_media_packets); RTC_DCHECK(fec_packets->empty()); @@ -239,7 +239,7 @@ void ForwardErrorCorrection::GenerateFecPayloads( 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 5339c656e5..a2d99e7143 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc @@ -50,8 +50,8 @@ bool RtcpPacket::OnBufferFull(uint8_t* packet, 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 a2d37a4dfc..322bf363b4 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::SetSubType(uint8_t subtype) { 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 cdb45b0d5f..494c8708f1 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 @@ bool Bye::Create(uint8_t* packet, *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 69df03a000..7f62e18c9b 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 @@ bool Fir::Create(uint8_t* packet, 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 ac5fc1a68f..ec7b51ad00 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 @@ bool Remb::Create(uint8_t* packet, 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 8015fa398b..e597748027 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 @@ bool ReportBlock::Parse(const uint8_t* buffer, size_t length) { 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::WriteBigEndian(&buffer[0], source_ssrc()); ByteWriter::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 a0b785c596..702afd8a8c 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 @@ bool Tmmbn::Create(uint8_t* packet, 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 b4b8ccba4e..0ba131187b 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 @@ bool Tmmbr::Create(uint8_t* packet, 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 9008aa0939..3a5a3038be 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 @@ class OneBitVectorChunk : public TransportFeedback::PacketStatusChunk { 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 b32e78ef9a..2344b2820c 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 @@ void RtpPacketizerH264::PacketizeFuA(size_t fragment_index) { 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 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) { 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 245ea8b352..72fd7892b2 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::SetSsrc(uint32_t ssrc) { } void Packet::SetCsrcs(const std::vector& 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 @@ void Packet::SetCsrcs(const std::vector& csrcs) { } 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 @@ uint8_t* Packet::AllocatePayload(size_t size_bytes) { } 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 @@ bool Packet::AllocateExtension(ExtensionType type, 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 0a15209271..c4edc732b8 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::SetStorePacketsStatus(bool enable, } 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 5842b90c0a..8f83e051ee 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 VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) { 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 ff7077e569..75e7043509 100644 --- a/webrtc/modules/video_coding/codec_database.cc +++ b/webrtc/modules/video_coding/codec_database.cc @@ -198,7 +198,7 @@ bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec, 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 b8a2dcd800..c5d2ed9e61 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 @@ bool Vp9FrameBufferPool::InitializeVpxUsePool( rtc::scoped_refptr Vp9FrameBufferPool::GetFrameBuffer(size_t min_size) { - RTC_DCHECK_GT(min_size, 0u); + RTC_DCHECK_GT(min_size, 0); rtc::scoped_refptr 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 f2aa6eabb6..f862faf0bb 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 57f9b268c0..c6db91619b 100644 --- a/webrtc/modules/video_coding/video_codec_initializer.cc +++ b/webrtc/modules/video_coding/video_codec_initializer.cc @@ -177,8 +177,8 @@ VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec( } 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 0d7975821f..d48e99066f 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 @@ bool H264CMSampleBufferToAnnexBBuffer( 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 H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer, 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 a2ffe99c14..71fefea72f 100644 --- a/webrtc/system_wrappers/include/aligned_array.h +++ b/webrtc/system_wrappers/include/aligned_array.h @@ -23,7 +23,7 @@ template class AlignedArray { 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(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 aaafdb3876..aadcb8d5b8 100644 --- a/webrtc/test/call_test.cc +++ b/webrtc/test/call_test.cc @@ -203,8 +203,8 @@ void CallTest::CreateSendConfig(size_t num_video_streams, 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 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { } } - 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 eeb0b0f9e5..b6307f94c1 100644 --- a/webrtc/test/frame_generator.cc +++ b/webrtc/test/frame_generator.cc @@ -161,7 +161,7 @@ class ScrollingImageFrameGenerator : public FrameGenerator { 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 cf8d5d099f..86c181bdf4 100644 --- a/webrtc/video/end_to_end_tests.cc +++ b/webrtc/video/end_to_end_tests.cc @@ -1039,7 +1039,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) { } // 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 @@ TEST_P(EndToEndTest, ReportsSetEncoderRates) { std::vector* 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 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx, 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 f7a63aea41..8f5cb367e2 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -157,7 +157,7 @@ class VideoAnalyzer : public PacketReceiver, // 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 @@ class VideoAnalyzer : public PacketReceiver, 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 @@ void VideoQualityTest::CheckParams() { 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 331f009287..0821059801 100644 --- a/webrtc/video/video_send_stream.cc +++ b/webrtc/video/video_send_stream.cc @@ -50,7 +50,7 @@ std::vector CreateRtpRtcpModules( 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 4412f4eaa5..359a01385a 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -1110,7 +1110,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) { VideoSendStream::Config* send_config, std::vector* 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 @@ class MaxPaddingSetTest : public test::SendTest { VideoSendStream::Config* send_config, std::vector* 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 1787915356..09ca1274c2 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 @@ bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) { } 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 37e12cea4f..88c60fdc8d 100644 --- a/webrtc/voice_engine/utility.cc +++ b/webrtc/voice_engine/utility.cc @@ -82,10 +82,10 @@ void MixWithSat(int16_t target[], 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 bac9dcadbf..fecb16a325 100644 --- a/webrtc/voice_engine/voe_base_impl.cc +++ b/webrtc/voice_engine/voe_base_impl.cc @@ -535,7 +535,7 @@ int VoEBaseImpl::GetVersion(char version[1024]) { } 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';