Format /modules

git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -e  "^modules/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jared Siskin <jtsiskin@meta.com>
Cr-Commit-Position: refs/heads/main@{#39901}

modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc

@@ -103,11 +103,7 @@ TEST(AudioDecoderFactoryTest, MaxNrOfChannels) {
 #ifdef WEBRTC_CODEC_ILBC
       "ilbc",
 #endif
-    "pcmu",
+      "pcmu", "pcma", "l16", "G722", "G711",
-    "pcma",
-    "l16",
-    "G722",
-    "G711",
   };
 
   for (auto codec : codecs) {

modules/audio_coding/codecs/ilbc/abs_quant.h

@@ -37,6 +37,6 @@ void WebRtcIlbcfix_AbsQuant(
                                    input) */
     int16_t* in,             /* (i) vector to encode */
     int16_t* weightDenum     /* (i) denominator of synthesis filter */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/bw_expand.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_BwExpand(
                                     expansion */
     int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
     int16_t length /* (i) the length of lpc coefficient vectors */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/cb_mem_energy.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_CbMemEnergy(
     int16_t* energyShifts, /* (o) Shift value of the energy */
     int scale,             /* (i) The scaling of all energy values */
     size_t base_size /* (i) Index to where energy values should be stored */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h

@@ -29,6 +29,6 @@ void WebRtcIlbcfix_CbMemEnergyAugmentation(
     size_t base_size,   /* (i) Index to where energy values should be stored */
     int16_t* energyW16, /* (o) Energy in the CB vectors */
     int16_t* energyShifts /* (o) Shift value of the energy */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_CbMemEnergyCalc(
     int16_t* energyShifts, /* (o) Shift value of the energy */
     int scale,             /* (i) The scaling of all energy values */
     size_t base_size /* (i) Index to where energy values should be stored */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/cb_search.h

@@ -35,6 +35,6 @@ void WebRtcIlbcfix_CbSearch(
     size_t lTarget,       /* (i) Length of vector */
     int16_t* weightDenum, /* (i) weighting filter coefficients in Q12 */
     size_t block          /* (i) the subblock number */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/chebyshev.h

@@ -33,6 +33,6 @@ int16_t WebRtcIlbcfix_Chebyshev(
     /* (o) Result of C(x) */
     int16_t x, /* (i) Value to the Chevyshev polynomial */
     int16_t* f /* (i) The coefficients in the polynomial */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/comp_corr.h

@@ -34,6 +34,6 @@ void WebRtcIlbcfix_CompCorr(int32_t* corr,   /* (o) cross correlation */
                             size_t bLen,     /* (i) length of buffer */
                             size_t sRange,   /* (i) correlation search length */
                             int16_t scale /* (i) number of rightshifts to use */
-                            );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h

@@ -36,6 +36,6 @@ void WebRtcIlbcfix_DecoderInterpolateLsp(
     int16_t length,       /* (i) length of lsf coefficient vector */
     IlbcDecoder* iLBCdec_inst
     /* (i) the decoder state structure */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/do_plc.h

@@ -39,6 +39,6 @@ void WebRtcIlbcfix_DoThePlc(
     size_t inlag,         /* (i) pitch lag */
     IlbcDecoder* iLBCdec_inst
     /* (i/o) decoder instance */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/encode.h

@@ -33,6 +33,6 @@ void WebRtcIlbcfix_EncodeImpl(
     const int16_t* block,     /* (i) speech vector to encode */
     IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder
                                            state */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/enh_upsample.h

@@ -28,6 +28,6 @@
 void WebRtcIlbcfix_EnhUpsample(
     int32_t* useq1, /* (o) upsampled output sequence */
     int16_t* seq1   /* (i) unupsampled sequence */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/enhancer.h

@@ -35,6 +35,6 @@ void WebRtcIlbcfix_Enhancer(
     size_t* period,        /* (i) pitch period array (pitch bward-in time) */
     const size_t* plocs,   /* (i) locations where period array values valid */
     size_t periodl         /* (i) dimension of period and plocs */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h

@@ -34,6 +34,6 @@ void WebRtcIlbcfix_FilteredCbVecs(
                                               second CB section */
     size_t lMem,        /* (i) Length of codebook memory */
     size_t samples      /* (i) Number of samples to filter */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/frame_classify.h

@@ -29,6 +29,6 @@ size_t WebRtcIlbcfix_FrameClassify(
     IlbcEncoder* iLBCenc_inst,
     /* (i/o) the encoder state structure */
     int16_t* residualFIX /* (i) lpc residual signal */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/gain_dequant.h

@@ -31,6 +31,6 @@ int16_t WebRtcIlbcfix_GainDequant(
     int16_t index, /* (i) quantization index */
     int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
     int16_t stage  /* (i) The stage of the search */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/gain_quant.h

@@ -31,6 +31,6 @@ WebRtcIlbcfix_GainQuant(               /* (o) quantized gain value */
                         int16_t maxIn, /* (i) maximum of gain value Q14 */
                         int16_t stage, /* (i) The stage of the search */
                         int16_t* index /* (o) quantization index */
-                        );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/get_sync_seq.h

@@ -36,6 +36,6 @@ void WebRtcIlbcfix_GetSyncSeq(
     size_t hl,           /* (i) 2*hl+1 is the number of sequences */
     int16_t* surround    /* (i/o) The contribution from this sequence
                                   summed with earlier contributions */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/index_conv_dec.h

@@ -22,6 +22,6 @@
 #include "modules/audio_coding/codecs/ilbc/defines.h"
 
 void WebRtcIlbcfix_IndexConvDec(int16_t* index /* (i/o) Codebook indexes */
-                                );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/index_conv_enc.h

@@ -26,6 +26,6 @@
  *---------------------------------------------------------------*/
 
 void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */
-                                );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/init_decode.h

@@ -33,6 +33,6 @@ int WebRtcIlbcfix_InitDecode(/* (o) Number of decoded samples */
                              int16_t mode,     /* (i) frame size mode */
                              int use_enhancer  /* (i) 1 to use enhancer
                                                   0 to run without enhancer */
-                             );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/init_encode.h

@@ -31,6 +31,6 @@ int WebRtcIlbcfix_InitEncode(/* (o) Number of bytes encoded */
                              IlbcEncoder*
                                  iLBCenc_inst, /* (i/o) Encoder instance */
                              int16_t mode      /* (i) frame size mode */
-                             );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/interpolate_samples.h

@@ -30,6 +30,6 @@ void WebRtcIlbcfix_InterpolateSamples(
     int16_t* interpSamples, /* (o) The interpolated samples */
     int16_t* CBmem,         /* (i) The CB memory */
     size_t lMem             /* (i) Length of the CB memory */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lpc_encode.h

@@ -37,6 +37,6 @@ void WebRtcIlbcfix_LpcEncode(
     int16_t* data,        /* (i) Speech to do LPC analysis on */
     IlbcEncoder* iLBCenc_inst
     /* (i/o) the encoder state structure */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_LspInterpolate2PolyDec(
     int16_t coef,  /* (i) weighting coefficient to use between
                                    lsf1 and lsf2 Q14 */
     int16_t length /* (i) length of coefficient vectors */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h

@@ -33,6 +33,6 @@ void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
     int16_t coef,  /* (i) weighting coefficient to use between
                             lsf1 and lsf2 Q14 */
     int16_t length /* (i) length of coefficient vectors */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lsf_to_lsp.h

@@ -29,6 +29,6 @@ void WebRtcIlbcfix_Lsf2Lsp(
     int16_t* lsf, /* (i) lsf in Q13 values between 0 and pi */
     int16_t* lsp, /* (o) lsp in Q15 values between -1 and 1 */
     int16_t m     /* (i) number of coefficients */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lsf_to_poly.h

@@ -28,6 +28,6 @@
 void WebRtcIlbcfix_Lsf2Poly(
     int16_t* a,  /* (o) predictor coefficients (order = 10) in Q12 */
     int16_t* lsf /* (i) line spectral frequencies in Q13 */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/lsp_to_lsf.h

@@ -30,6 +30,6 @@ void WebRtcIlbcfix_Lsp2Lsf(
     int16_t* lsf, /* (o) Lsf vector 0...Pi in Q13
                            (ordered, so that lsf[i]<lsf[i+1]) */
     int16_t m     /* (i) Number of coefficients */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/my_corr.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_MyCorr(int32_t* corr, /* (o) correlation of seq1 and seq2 */
                           size_t dim1,         /* (i) dimension first seq1 */
                           const int16_t* seq2, /* (i) second sequence */
                           size_t dim2          /* (i) dimension seq2 */
-                          );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/nearest_neighbor.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_NearestNeighbor(
     const size_t* array, /* (i) data array (Q2) */
     size_t value,        /* (i) value (Q2) */
     size_t arlength      /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/pack_bits.h

@@ -29,6 +29,6 @@ void WebRtcIlbcfix_PackBits(
     uint16_t* bitstream, /* (o) The packetized bitstream */
     iLBC_bits* enc_bits, /* (i) Encoded bits */
     int16_t mode         /* (i) Codec mode (20 or 30) */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/poly_to_lsf.h

@@ -27,6 +27,6 @@
 
 void WebRtcIlbcfix_Poly2Lsf(int16_t* lsf, /* (o) lsf coefficients (Q13) */
                             int16_t* a    /* (i) A coefficients (Q12) */
-                            );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/poly_to_lsp.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_Poly2Lsp(
     int16_t* lsp,    /* (i) LSP coefficients in Q15 */
     int16_t* old_lsp /* (i) old LSP coefficients that are used if the new
                               coefficients turn out to be unstable */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/refiner.h

@@ -39,6 +39,6 @@ void WebRtcIlbcfix_Refiner(
     int16_t* surround,     /* (i/o) The contribution from this sequence
                                     summed with earlier contributions */
     int16_t gain           /* (i) Gain to use for this sequence */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h

@@ -43,6 +43,6 @@ void WebRtcIlbcfix_SimpleInterpolateLsf(
     int16_t length,       /* (i) should equate FILTERORDER */
     IlbcEncoder* iLBCenc_inst
     /* (i/o) the encoder state structure */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_SimpleLpcAnalysis(
     int16_t* data, /* (i) new block of speech */
     IlbcEncoder* iLBCenc_inst
     /* (i/o) the encoder state structure */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h

@@ -29,6 +29,6 @@ void WebRtcIlbcfix_SimpleLsfDeQ(
     int16_t* lsfdeq, /* (o) dequantized lsf coefficients */
     int16_t* index,  /* (i) quantization index */
     int16_t lpc_n    /* (i) number of LPCs */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/simple_lsf_quant.h

@@ -32,6 +32,6 @@ void WebRtcIlbcfix_SimpleLsfQ(
     int16_t* lsf,    /* (i) the lsf coefficient vector to be
                               quantized (dimension FILTERORDER) Q13 */
     int16_t lpc_n    /* (i) number of lsf sets to quantize */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/smooth.h

@@ -30,6 +30,6 @@ void WebRtcIlbcfix_Smooth(int16_t* odata,   /* (o) smoothed output */
                                                      this block */
                           int16_t* surround /* (i) The approximation from the
                                                      surrounding sequences */
-                          );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/sort_sq.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_SortSq(
     int16_t x,         /* (i) the value to quantize */
     const int16_t* cb, /* (i) the quantization codebook */
     int16_t cb_size    /* (i) the size of the quantization codebook */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/split_vq.h

@@ -33,6 +33,6 @@ void WebRtcIlbcfix_SplitVq(
     int16_t* CB,    /* (i) the quantizer codebook in Q13 */
     int16_t* dim,   /* (i) the dimension of X and qX */
     int16_t* cbsize /* (i) the number of vectors in the codebook */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/state_construct.h

@@ -33,6 +33,6 @@ void WebRtcIlbcfix_StateConstruct(
     int16_t* syntDenum, /* (i) synthesis filter denumerator */
     int16_t* Out_fix,   /* (o) the decoded state vector */
     size_t len          /* (i) length of a state vector */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/state_search.h

@@ -36,6 +36,6 @@ void WebRtcIlbcfix_StateSearch(
     int16_t* residual,       /* (i) target residual vector */
     int16_t* syntDenum,      /* (i) lpc synthesis filter */
     int16_t* weightDenum     /* (i) weighting filter denuminator */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/swap_bytes.h

@@ -30,6 +30,6 @@ void WebRtcIlbcfix_SwapBytes(
     const uint16_t* input, /* (i) the sequence to swap */
     size_t wordLength,     /* (i) number or uint16_t to swap */
     uint16_t* output       /* (o) the swapped sequence */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/unpack_bits.h

@@ -34,6 +34,6 @@ WebRtcIlbcfix_UnpackBits(/* (o) "Empty" frame indicator */
                          iLBC_bits*
                              enc_bits, /* (o) Paramerers from bitstream */
                          int16_t mode  /* (i) Codec mode (20 or 30) */
-                         );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/vq3.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_Vq3(
     int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
     int16_t* X,     /* (i) the vector to quantize (Q13) */
     int16_t n_cb    /* (i) the number of vectors in the codebook */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/vq4.h

@@ -31,6 +31,6 @@ void WebRtcIlbcfix_Vq4(
     int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
     int16_t* X,     /* (i) the vector to quantize (Q13) */
     int16_t n_cb    /* (i) the number of vectors in the codebook */
-    );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/window32_w32.h

@@ -30,6 +30,6 @@ void WebRtcIlbcfix_Window32W32(int32_t* z, /* Output */
                                int32_t* x, /* Input (same domain as Output)*/
                                const int32_t* y, /* Q31 Window */
                                size_t N          /* length to process */
-                               );
+);
 
 #endif

modules/audio_coding/codecs/ilbc/xcorr_coef.h

@@ -34,6 +34,6 @@ size_t WebRtcIlbcfix_XcorrCoef(
     size_t searchLen,   /* (i) the search lenght */
     size_t offset,      /* (i) samples offset between arrays */
     int16_t step        /* (i) +1 or -1 */
-    );
+);
 
 #endif

modules/audio_coding/codecs/opus/opus_interface.cc

@@ -11,7 +11,6 @@
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 
 #include <cstdlib>
-
 #include <numeric>
 
 #include "api/array_view.h"

modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc

@@ -37,7 +37,7 @@ namespace {
 static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
 static const size_t kRedLastHeaderLength =
     1;  // 1 byte RED header for the last element.
-}
+}  // namespace
 
 class AudioEncoderCopyRedTest : public ::testing::Test {
  protected:

modules/audio_coding/neteq/accelerate.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/neteq/accelerate.h"
 
-
 #include "api/array_view.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 

modules/audio_coding/neteq/audio_multi_vector.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 
-
 #include <algorithm>
 
 #include "rtc_base/checks.h"

modules/audio_coding/neteq/audio_vector.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/neteq/audio_vector.h"
 
-
 #include <algorithm>
 #include <memory>
 

modules/audio_coding/neteq/buffer_level_filter.cc

@@ -35,7 +35,7 @@ void BufferLevelFilter::Update(size_t buffer_size_samples,
   // `level_factor_` and `filtered_current_level_` are in Q8.
   // `buffer_size_samples` is in Q0.
   const int64_t filtered_current_level =
-      (level_factor_ * int64_t{filtered_current_level_} >> 8) +
+      (level_factor_* int64_t{filtered_current_level_} >> 8) +
       (256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);
 
   // Account for time-scale operations (accelerate and pre-emptive expand) and

modules/audio_coding/neteq/comfort_noise.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/neteq/comfort_noise.h"
 
-
 #include <cstdint>
 #include <memory>
 

modules/audio_coding/neteq/expand.h

@@ -11,7 +11,6 @@
 #ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
 #define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
 
-
 #include <memory>
 
 #include "modules/audio_coding/neteq/audio_vector.h"

modules/audio_coding/neteq/neteq_unittest.cc

@@ -160,7 +160,6 @@ TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
   EXPECT_EQ(-1, stats.max_waiting_time_ms);
 }
 
-
 TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
   // Apply a clock drift of -25 ms / s (sender faster than receiver).
   const double kDriftFactor = 1000.0 / (1000.0 + 25.0);

modules/audio_coding/neteq/red_payload_splitter_unittest.cc

@@ -12,7 +12,6 @@
 
 #include "modules/audio_coding/neteq/red_payload_splitter.h"
 
-
 #include <memory>
 #include <utility>  // pair
 

modules/audio_coding/neteq/tools/neteq_test.cc

@@ -313,8 +313,7 @@ NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
 }
 
 NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
-  DecoderMap codecs = {
+  DecoderMap codecs = {{0, SdpAudioFormat("pcmu", 8000, 1)},
-    {0, SdpAudioFormat("pcmu", 8000, 1)},
                        {8, SdpAudioFormat("pcma", 8000, 1)},
 #ifdef WEBRTC_CODEC_ILBC
                        {102, SdpAudioFormat("ilbc", 8000, 1)},
@@ -335,8 +334,7 @@ NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
                        {13, SdpAudioFormat("cn", 8000, 1)},
                        {98, SdpAudioFormat("cn", 16000, 1)},
                        {99, SdpAudioFormat("cn", 32000, 1)},
-    {100, SdpAudioFormat("cn", 48000, 1)}
+                       {100, SdpAudioFormat("cn", 48000, 1)}};
-  };
   return codecs;
 }
 

modules/audio_coding/neteq/tools/rtp_file_source.cc

@@ -80,8 +80,7 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
 }
 
 RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
-    : PacketSource(),
+    : PacketSource(), ssrc_filter_(ssrc_filter) {}
-      ssrc_filter_(ssrc_filter) {}
 
 bool RtpFileSource::OpenFile(absl::string_view file_name) {
   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));

modules/audio_coding/neteq/tools/rtp_generator.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_coding/neteq/tools/rtp_generator.h"
 
-
 namespace webrtc {
 namespace test {
 

modules/audio_device/linux/audio_device_alsa_linux.cc

@@ -10,7 +10,6 @@
 
 #include "modules/audio_device/linux/audio_device_alsa_linux.h"
 
-
 #include "modules/audio_device/audio_device_config.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/system/arch.h"

modules/audio_device/win/audio_device_core_win.cc

@@ -29,12 +29,11 @@
 #include "modules/audio_device/win/audio_device_core_win.h"
 // clang-format on
 
-#include <string.h>
-
 #include <comdef.h>
 #include <dmo.h>
 #include <functiondiscoverykeys_devpkey.h>
 #include <mmsystem.h>
+#include <string.h>
 #include <strsafe.h>
 #include <uuids.h>
 #include <windows.h>
@@ -3256,8 +3255,9 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() {
         QueryPerformanceCounter(&t1);
 
         // Get the current recording and playout delay.
-        uint32_t sndCardRecDelay = (uint32_t)(
+        uint32_t sndCardRecDelay =
-            ((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) / 10000) +
+            (uint32_t)(((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) /
+                        10000) +
                        (10 * syncBufIndex) / _recBlockSize - 10);
         uint32_t sndCardPlayDelay = static_cast<uint32_t>(_sndCardPlayDelay);
 

modules/audio_device/win/audio_device_core_win.h

@@ -13,12 +13,9 @@
 
 #if (_MSC_VER >= 1400)  // only include for VS 2005 and higher
 
-#include "rtc_base/win32.h"
-
-#include "modules/audio_device/audio_device_generic.h"
-
 #include <wmcodecdsp.h>  // CLSID_CWMAudioAEC
-                          // (must be before audioclient.h)
+//(must be before audioclient.h)
+
 #include <audioclient.h>  // WASAPI
 #include <audiopolicy.h>
 #include <avrt.h>  // Avrt
@@ -27,8 +24,10 @@
 #include <mmdeviceapi.h>  // MMDevice
 
 #include "api/scoped_refptr.h"
+#include "modules/audio_device/audio_device_generic.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/win/scoped_com_initializer.h"
+#include "rtc_base/win32.h"
 
 // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
 #pragma comment(lib, "avrt.lib")

modules/audio_device/win/core_audio_utility_win_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "modules/audio_device/win/core_audio_utility_win.h"
+
 #include "rtc_base/arraysize.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/win/scoped_com_initializer.h"

modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc

@@ -8,10 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
-
 #include <immintrin.h>
 
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
-
 #include <immintrin.h>
 
+#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h"
+
 namespace webrtc {
 
 namespace aec3 {

modules/audio_processing/aec3/echo_path_variability.h

@@ -14,11 +14,7 @@
 namespace webrtc {
 
 struct EchoPathVariability {
-  enum class DelayAdjustment {
+  enum class DelayAdjustment { kNone, kBufferFlush, kNewDetectedDelay };
-    kNone,
-    kBufferFlush,
-    kNewDetectedDelay
-  };
 
   EchoPathVariability(bool gain_change,
                       DelayAdjustment delay_change,

modules/audio_processing/aec3/fft_data_avx2.cc

@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/fft_data.h"
-
 #include <immintrin.h>
 
 #include "api/array_view.h"
+#include "modules/audio_processing/aec3/fft_data.h"
 
 namespace webrtc {
 

modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc

@@ -453,18 +453,21 @@ TEST(MultiChannelContentDetectorMetrics, ReportsMetrics) {
                              "PersistentMultichannelContentEverDetected"));
   EXPECT_METRIC_EQ(
       1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
-                             "PersistentMultichannelContentEverDetected", 1));
+                            "PersistentMultichannelContentEverDetected",
+                            1));
 
   // Check periodic metric.
   EXPECT_METRIC_EQ(
       2, metrics::NumSamples("WebRTC.Audio.EchoCanceller."
                              "ProcessingPersistentMultichannelContent"));
-  EXPECT_METRIC_EQ(
+  EXPECT_METRIC_EQ(1,
-      1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+                   metrics::NumEvents("WebRTC.Audio.EchoCanceller."
-                             "ProcessingPersistentMultichannelContent", 0));
+                                      "ProcessingPersistentMultichannelContent",
-  EXPECT_METRIC_EQ(
+                                      0));
-      1, metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+  EXPECT_METRIC_EQ(1,
-                             "ProcessingPersistentMultichannelContent", 1));
+                   metrics::NumEvents("WebRTC.Audio.EchoCanceller."
+                                      "ProcessingPersistentMultichannelContent",
+                                      1));
 }
 
 }  // namespace webrtc

modules/audio_processing/aec3/reverb_model.h

@@ -49,7 +49,6 @@ class ReverbModel {
                     float reverb_decay);
 
  private:
-
   std::array<float, kFftLengthBy2Plus1> reverb_;
 };
 

modules/audio_processing/aec3/vector_math_avx2.cc

@@ -8,12 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/aec3/vector_math.h"
-
 #include <immintrin.h>
 #include <math.h>
 
 #include "api/array_view.h"
+#include "modules/audio_processing/aec3/vector_math.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

modules/audio_processing/aecm/aecm_core.cc

@@ -123,7 +123,6 @@ const int16_t WebRtcAecm_kSinTable[] = {
     -2667, -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
     -1140, -998,  -856,  -713,  -571,  -428,  -285,  -142};
 
-
 // Moves the pointer to the next entry and inserts `far_spectrum` and
 // corresponding Q-domain in its buffer.
 //

modules/audio_processing/aecm/aecm_core_c.cc

@@ -185,8 +185,9 @@ static void WindowAndFFT(AecmCore* aecm,
     int16_t scaled_time_signal = time_signal[i] * (1 << time_signal_scaling);
     fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14);
     scaled_time_signal = time_signal[i + PART_LEN] * (1 << time_signal_scaling);
-    fft[PART_LEN + i] = (int16_t)(
+    fft[PART_LEN + i] = (int16_t)((scaled_time_signal *
-        (scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14);
+                                   WebRtcAecm_kSqrtHanning[PART_LEN - i]) >>
+                                  14);
   }
 
   // Do forward FFT, then take only the first PART_LEN complex samples,
@@ -644,18 +645,18 @@ int RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/8200
       }
 
       // multiply with Wiener coefficients
-      efw[i].real = (int16_t)(
+      efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+                                                                   hnl[i], 14));
-      efw[i].imag = (int16_t)(
+      efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+                                                                   hnl[i], 14));
     }
   } else {
     // multiply with Wiener coefficients
     for (i = 0; i < PART_LEN1; i++) {
-      efw[i].real = (int16_t)(
+      efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+                                                                   hnl[i], 14));
-      efw[i].imag = (int16_t)(
+      efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+                                                                   hnl[i], 14));
     }
   }
 

modules/audio_processing/aecm/aecm_core_mips.cc

@@ -569,8 +569,8 @@ static void InverseFFTAndWindow(AecmCore* aecm,
         [paecm_buf] "+r"(paecm_buf), [i] "=&r"(i),
         [pp_kSqrtHanning] "+r"(pp_kSqrtHanning),
         [p_kSqrtHanning] "+r"(p_kSqrtHanning)
-      : [out_aecm] "r"(out_aecm),
+      : [out_aecm] "r"(out_aecm), [WebRtcAecm_kSqrtHanning] "r"(
-        [WebRtcAecm_kSqrtHanning] "r"(WebRtcAecm_kSqrtHanning)
+                                      WebRtcAecm_kSqrtHanning)
       : "hi", "lo", "memory");
 
   // Copy the current block to the old position
@@ -1334,10 +1334,10 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
   } else {
     // multiply with Wiener coefficients
     for (i = 0; i < PART_LEN1; i++) {
-      efw[i].real = (int16_t)(
+      efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
+                                                                   hnl[i], 14));
-      efw[i].imag = (int16_t)(
+      efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
-          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
+                                                                   hnl[i], 14));
     }
   }
 
@@ -1424,8 +1424,8 @@ static void ComfortNoise(AecmCore* aecm,
             "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
             "subu   %[tnoise],      %[tnoise],      %[tmp32]            \n\t"
             : [tmp32] "=&r"(tmp32), [tnoise] "+r"(tnoise)
-            :
+            : [outLShift32] "r"(outLShift32), [minTrackShift] "r"(
-            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
+                                                  minTrackShift));
       }
     } else {
       // Reset "too high" counter
@@ -1497,8 +1497,8 @@ static void ComfortNoise(AecmCore* aecm,
             "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
             "subu   %[tnoise1],     %[tnoise1],     %[tmp32]            \n\t"
             : [tmp32] "=&r"(tmp32), [tnoise1] "+r"(tnoise1)
-            :
+            : [outLShift32] "r"(outLShift32), [minTrackShift] "r"(
-            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
+                                                  minTrackShift));
       }
     } else {
       // Reset "too high" counter

modules/audio_processing/agc/legacy/analog_agc.h

@@ -11,7 +11,6 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
 #define MODULES_AUDIO_PROCESSING_AGC_LEGACY_ANALOG_AGC_H_
 
-
 #include "modules/audio_processing/agc/legacy/digital_agc.h"
 #include "modules/audio_processing/agc/legacy/gain_control.h"
 

modules/audio_processing/agc2/rnn_vad/rnn_fc.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
+
 #include <algorithm>
 #include <numeric>
 
-#include "modules/audio_processing/agc2/rnn_vad/rnn_fc.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "third_party/rnnoise/src/rnn_activations.h"

modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc

@@ -8,11 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
-
 #include <immintrin.h>
 
 #include "api/array_view.h"
+#include "modules/audio_processing/agc2/rnn_vad/vector_math.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 

modules/audio_processing/gain_control_impl.cc

@@ -255,7 +255,6 @@ int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
   return AudioProcessing::kNoError;
 }
 
-
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
   data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level);
@@ -287,7 +286,6 @@ int GainControlImpl::set_mode(Mode mode) {
   return AudioProcessing::kNoError;
 }
 
-
 int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
   if (minimum < 0 || maximum > 65535 || maximum < minimum) {
     return AudioProcessing::kBadParameterError;
@@ -302,7 +300,6 @@ int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
   return AudioProcessing::kNoError;
 }
 
-
 int GainControlImpl::set_target_level_dbfs(int level) {
   if (level > 31 || level < 0) {
     return AudioProcessing::kBadParameterError;

modules/audio_processing/ns/noise_suppressor.cc

@@ -13,6 +13,7 @@
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

modules/audio_processing/ns/prior_signal_model_estimator.cc

@@ -11,6 +11,7 @@
 #include "modules/audio_processing/ns/prior_signal_model_estimator.h"
 
 #include <math.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

modules/audio_processing/ns/quantile_noise_estimator.h

@@ -12,6 +12,7 @@
 #define MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_
 
 #include <math.h>
+
 #include <array>
 
 #include "api/array_view.h"

modules/audio_processing/ns/speech_probability_estimator.cc

@@ -11,6 +11,7 @@
 #include "modules/audio_processing/ns/speech_probability_estimator.h"
 
 #include <math.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

modules/audio_processing/ns/wiener_filter.cc

@@ -13,6 +13,7 @@
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
+
 #include <algorithm>
 
 #include "modules/audio_processing/ns/fast_math.h"

modules/audio_processing/test/conversational_speech/generator.cc

@@ -9,9 +9,8 @@
  */
 
 #include <iostream>
-#include <vector>
-
 #include <memory>
+#include <vector>
 
 #include "absl/flags/flag.h"
 #include "absl/flags/parse.h"

modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc

@@ -968,9 +968,7 @@ TEST(GoogCcScenario, FallbackToLossBasedBweWithoutPacketFeedback) {
   EXPECT_GE(client->target_rate().kbps(), 500);
 
   // Update the network to create high loss ratio
-  net->UpdateConfig([](NetworkSimulationConfig* c) {
+  net->UpdateConfig([](NetworkSimulationConfig* c) { c->loss_rate = 0.15; });
-    c->loss_rate = 0.15;
-  });
   s.RunFor(TimeDelta::Seconds(20));
 
   // Bandwidth decreases thanks to loss based bwe v0.

modules/congestion_controller/pcc/bitrate_controller.cc

@@ -17,7 +17,6 @@
 #include <utility>
 #include <vector>
 
-
 namespace webrtc {
 namespace pcc {
 

modules/congestion_controller/rtp/transport_feedback_adapter.cc

@@ -85,7 +85,6 @@ bool InFlightBytesTracker::NetworkRouteComparator::operator()(
 
 TransportFeedbackAdapter::TransportFeedbackAdapter() = default;
 
-
 void TransportFeedbackAdapter::AddPacket(const RtpPacketSendInfo& packet_info,
                                          size_t overhead_bytes,
                                          Timestamp creation_time) {
@@ -213,13 +212,13 @@ TransportFeedbackAdapter::ProcessTransportFeedbackInner(
   size_t failed_lookups = 0;
   size_t ignored = 0;
 
-  feedback.ForAllPackets([&](uint16_t sequence_number,
+  feedback.ForAllPackets(
-                             TimeDelta delta_since_base) {
+      [&](uint16_t sequence_number, TimeDelta delta_since_base) {
         int64_t seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
 
         if (seq_num > last_ack_seq_num_) {
-      // Starts at history_.begin() if last_ack_seq_num_ < 0, since any valid
+          // Starts at history_.begin() if last_ack_seq_num_ < 0, since any
-      // sequence number is >= 0.
+          // valid sequence number is >= 0.
           for (auto it = history_.upper_bound(last_ack_seq_num_);
                it != history_.upper_bound(seq_num); ++it) {
             in_flight_.RemoveInFlightPacketBytes(it->second);
@@ -244,9 +243,10 @@ TransportFeedbackAdapter::ProcessTransportFeedbackInner(
         PacketFeedback packet_feedback = it->second;
         if (delta_since_base.IsFinite()) {
           packet_feedback.receive_time =
-          current_offset_ + delta_since_base.RoundDownTo(TimeDelta::Millis(1));
+              current_offset_ +
-      // Note: Lost packets are not removed from history because they might be
+              delta_since_base.RoundDownTo(TimeDelta::Millis(1));
-      // reported as received by a later feedback.
+          // Note: Lost packets are not removed from history because they might
+          // be reported as received by a later feedback.
           history_.erase(it);
         }
         if (packet_feedback.network_route == network_route_) {

modules/congestion_controller/rtp/transport_feedback_demuxer.cc

@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+
 #include "absl/algorithm/container.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 

modules/desktop_capture/desktop_frame_unittest.cc

@@ -44,8 +44,8 @@ void RunTest(const TestData& test) {
   auto dest_frame = CreateTestFrame(test.dest_frame_rect, 0);
   auto src_frame = CreateTestFrame(test.src_frame_rect, 0xff);
 
-  dest_frame->CopyIntersectingPixelsFrom(
+  dest_frame->CopyIntersectingPixelsFrom(*src_frame, test.horizontal_scale,
-      *src_frame, test.horizontal_scale, test.vertical_scale);
+                                         test.vertical_scale);
 
   // Translate the expected overlap rect to be relative to the dest frame/rect.
   DesktopVector dest_frame_origin = test.dest_frame_rect.top_left();
@@ -105,6 +105,7 @@ TEST(DesktopFrameTest, FrameDataSwitchesBetweenNonBlackAndBlack) {
 }
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRects) {
+  // clang-format off
   const TestData tests[] = {
     {"0 origin",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -118,6 +119,7 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRects) {
      1.0, 1.0,
      DesktopRect::MakeXYWH(-1, -1, 2, 2)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
@@ -125,6 +127,7 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRects) {
 TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRectsScaled) {
   // The scale factors shouldn't affect matching rects (they're only applied
   // to any difference between the origins)
+  // clang-format off
   const TestData tests[] = {
     {"0 origin 2x",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -150,11 +153,13 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsMatchingRectsScaled) {
      0.5, 0.5,
      DesktopRect::MakeXYWH(-1, -1, 2, 2)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRects) {
+  // clang-format off
   const TestData tests[] = {
     {"0 origin top left",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -174,11 +179,13 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRects) {
      1.0, 1.0,
      DesktopRect::MakeXYWH(-1, 0, 1, 1)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRectsScaled) {
+  // clang-format off
   const TestData tests[] = {
     {"0 origin top left 2x",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -222,12 +229,14 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsFullyContainedRectsScaled) {
      0.5, 0.5,
      DesktopRect::MakeXYWH(-1, -1, 1, 1)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRects) {
+  // clang-format off
   const TestData tests[] = {
     {"Top left",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -253,11 +262,13 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRects) {
      1.0, 1.0,
      DesktopRect::MakeXYWH(0, 1, 1, 1)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRectsScaled) {
+  // clang-format off
   const TestData tests[] = {
     {"Top left 2x",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -283,12 +294,14 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsPartiallyContainedRectsScaled) {
      0.5, 0.5,
      DesktopRect::MakeXYWH(0, 1, 1, 1)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRects) {
+  // clang-format off
   const TestData tests[] = {
     {"Left",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -315,11 +328,13 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRects) {
      1.0, 1.0,
      DesktopRect::MakeXYWH(0, 0, 0, 0)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }
 
 TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRectsScaled) {
+  // clang-format off
   const TestData tests[] = {
     {"Left 2x",
      DesktopRect::MakeXYWH(0, 0, 2, 2),
@@ -346,6 +361,7 @@ TEST(DesktopFrameTest, CopyIntersectingPixelsUncontainedRectsScaled) {
      0.5, 0.5,
      DesktopRect::MakeXYWH(0, 0, 0, 0)}
   };
+  // clang-format on
 
   RunTests(tests, arraysize(tests));
 }

modules/desktop_capture/full_screen_application_handler.cc

@@ -9,6 +9,7 @@
  */
 
 #include "modules/desktop_capture/full_screen_application_handler.h"
+
 #include "rtc_base/logging.h"
 
 namespace webrtc {

modules/desktop_capture/full_screen_window_detector.cc

@@ -9,6 +9,7 @@
  */
 
 #include "modules/desktop_capture/full_screen_window_detector.h"
+
 #include "modules/desktop_capture/full_screen_application_handler.h"
 #include "rtc_base/time_utils.h"
 

modules/desktop_capture/linux/wayland/screencast_portal.cc

@@ -24,12 +24,12 @@ namespace {
 using xdg_portal::kScreenCastInterfaceName;
 using xdg_portal::PrepareSignalHandle;
 using xdg_portal::RequestResponse;
+using xdg_portal::RequestResponseFromPortalResponse;
 using xdg_portal::RequestSessionProxy;
 using xdg_portal::SetupRequestResponseSignal;
 using xdg_portal::SetupSessionRequestHandlers;
 using xdg_portal::StartSessionRequest;
 using xdg_portal::TearDownSession;
-using xdg_portal::RequestResponseFromPortalResponse;
 
 }  // namespace
 

modules/desktop_capture/linux/wayland/shared_screencast_stream.cc

@@ -449,8 +449,8 @@ bool SharedScreenCastStreamPrivate::StartScreenCastStream(
     PipeWireThreadLoopLock thread_loop_lock(pw_main_loop_);
 
     if (fd >= 0) {
-      pw_core_ = pw_context_connect_fd(
+      pw_core_ = pw_context_connect_fd(pw_context_, fcntl(fd, F_DUPFD_CLOEXEC),
-          pw_context_, fcntl(fd, F_DUPFD_CLOEXEC), nullptr, 0);
+                                       nullptr, 0);
     } else {
       pw_core_ = pw_context_connect(pw_context_, nullptr, 0);
     }

modules/desktop_capture/linux/x11/x_error_trap.cc

@@ -10,10 +10,10 @@
 
 #include "modules/desktop_capture/linux/x11/x_error_trap.h"
 
-#include <atomic>
-
 #include <stddef.h>
 
+#include <atomic>
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {

modules/desktop_capture/mac/full_screen_mac_application_handler.h

@@ -12,6 +12,7 @@
 #define MODULES_DESKTOP_CAPTURE_MAC_FULL_SCREEN_MAC_APPLICATION_HANDLER_H_
 
 #include <memory>
+
 #include "modules/desktop_capture/full_screen_application_handler.h"
 
 namespace webrtc {

modules/desktop_capture/mac/screen_capturer_mac.h

@@ -36,8 +36,7 @@ class DisplayStreamManager;
 // A class to perform video frame capturing for mac.
 class ScreenCapturerMac final : public DesktopCapturer {
  public:
-  ScreenCapturerMac(
+  ScreenCapturerMac(rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
-      rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
                     bool detect_updated_region,
                     bool allow_iosurface);
   ~ScreenCapturerMac() override;