diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc index 4a2b261a59..bd8d1cc341 100644 --- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc +++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc @@ -98,16 +98,12 @@ TEST(AudioDecoderFactoryTest, MaxNrOfChannels) { CreateBuiltinAudioDecoderFactory(); std::vector codecs = { #ifdef WEBRTC_CODEC_OPUS - "opus", + "opus", #endif #ifdef WEBRTC_CODEC_ILBC - "ilbc", + "ilbc", #endif - "pcmu", - "pcma", - "l16", - "G722", - "G711", + "pcmu", "pcma", "l16", "G722", "G711", }; for (auto codec : codecs) { diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc index 1ecc9bc3d1..e969ed1189 100644 --- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc +++ b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc @@ -94,7 +94,7 @@ int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded, const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) * Channels() / 2; // 1/2 byte per sample per channel - int16_t temp_type = 1; // Default is speech. + int16_t temp_type = 1; // Default is speech. // De-interleave the bit-stream into two separate payloads. uint8_t* encoded_deinterleaved = new uint8_t[encoded_len_adjusted]; SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved); diff --git a/modules/audio_coding/codecs/ilbc/abs_quant.h b/modules/audio_coding/codecs/ilbc/abs_quant.h index c72e29cf29..4a3f004ed3 100644 --- a/modules/audio_coding/codecs/ilbc/abs_quant.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/bw_expand.h b/modules/audio_coding/codecs/ilbc/bw_expand.h index ff9b0b302e..022c113dda 100644 --- a/modules/audio_coding/codecs/ilbc/bw_expand.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy.h index 17ec337dc6..15dc884f2a 100644 --- a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h index d7b7a0d97e..c489ab54f9 100644 --- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h index 1d1e8d62b9..4b3703182e 100644 --- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/cb_search.h b/modules/audio_coding/codecs/ilbc/cb_search.h index 84a52c7868..11856649e7 100644 --- a/modules/audio_coding/codecs/ilbc/cb_search.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/cb_search_core.h b/modules/audio_coding/codecs/ilbc/cb_search_core.h index 5da70e0988..5a3b13e446 100644 --- a/modules/audio_coding/codecs/ilbc/cb_search_core.h +++ b/modules/audio_coding/codecs/ilbc/cb_search_core.h @@ -33,9 +33,9 @@ void WebRtcIlbcfix_CbSearchCore( size_t* bestIndex, /* (o) Index that corresponds to maximum criteria (in this vector) */ - int32_t* bestCrit, /* (o) Value of critera for the - chosen index */ - int16_t* bestCritSh); /* (o) The domain of the chosen - criteria */ + int32_t* bestCrit, /* (o) Value of critera for the + chosen index */ + int16_t* bestCritSh); /* (o) The domain of the chosen + criteria */ #endif diff --git a/modules/audio_coding/codecs/ilbc/chebyshev.h b/modules/audio_coding/codecs/ilbc/chebyshev.h index 7e7742c5cc..8ba82927b8 100644 --- a/modules/audio_coding/codecs/ilbc/chebyshev.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/comp_corr.h b/modules/audio_coding/codecs/ilbc/comp_corr.h index 010c6a1ce5..d9df9a78f8 100644 --- a/modules/audio_coding/codecs/ilbc/comp_corr.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h index d7e5be1c2f..5bed469a12 100644 --- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h +++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h @@ -28,8 +28,8 @@ *----------------------------------------------------------------*/ void WebRtcIlbcfix_CreateAugmentedVec( - size_t index, /* (i) Index for the augmented vector to be - created */ + size_t index, /* (i) Index for the augmented vector to be + created */ const int16_t* buffer, /* (i) Pointer to the end of the codebook memory that is used for creation of the augmented codebook */ diff --git a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h index 8b08114467..40510007a9 100644 --- a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h +++ b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h @@ -29,13 +29,13 @@ *---------------------------------------------------------------*/ void WebRtcIlbcfix_DecoderInterpolateLsp( - int16_t* syntdenum, /* (o) synthesis filter coefficients */ + int16_t* syntdenum, /* (o) synthesis filter coefficients */ int16_t* weightdenum, /* (o) weighting denumerator coefficients */ - int16_t* lsfdeq, /* (i) dequantized lsf coefficients */ - int16_t length, /* (i) length of lsf coefficient vector */ + int16_t* lsfdeq, /* (i) dequantized lsf coefficients */ + int16_t length, /* (i) length of lsf coefficient vector */ IlbcDecoder* iLBCdec_inst /* (i) the decoder state structure */ - ); +); #endif diff --git a/modules/audio_coding/codecs/ilbc/do_plc.h b/modules/audio_coding/codecs/ilbc/do_plc.h index c19c4eca32..5e3bcc6d3c 100644 --- a/modules/audio_coding/codecs/ilbc/do_plc.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/encode.h b/modules/audio_coding/codecs/ilbc/encode.h index bc3e187d92..5290420bbf 100644 --- a/modules/audio_coding/codecs/ilbc/encode.h +++ b/modules/audio_coding/codecs/ilbc/encode.h @@ -29,10 +29,10 @@ *---------------------------------------------------------------*/ void WebRtcIlbcfix_EncodeImpl( - uint16_t* bytes, /* (o) encoded data bits iLBC */ - const int16_t* block, /* (i) speech vector to encode */ + uint16_t* bytes, /* (o) encoded data bits iLBC */ + const int16_t* block, /* (i) speech vector to encode */ IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder state */ - ); +); #endif diff --git a/modules/audio_coding/codecs/ilbc/energy_inverse.h b/modules/audio_coding/codecs/ilbc/energy_inverse.h index 15391cf230..3a11488056 100644 --- a/modules/audio_coding/codecs/ilbc/energy_inverse.h +++ b/modules/audio_coding/codecs/ilbc/energy_inverse.h @@ -28,8 +28,8 @@ void WebRtcIlbcfix_EnergyInverse( int16_t* - energy, /* (i/o) Energy and inverse - energy (in Q29) */ + energy, /* (i/o) Energy and inverse + energy (in Q29) */ size_t noOfEnergies); /* (i) The length of the energy vector */ diff --git a/modules/audio_coding/codecs/ilbc/enh_upsample.h b/modules/audio_coding/codecs/ilbc/enh_upsample.h index b427eca50a..20c85fb20e 100644 --- a/modules/audio_coding/codecs/ilbc/enh_upsample.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/enhancer.h b/modules/audio_coding/codecs/ilbc/enhancer.h index 386949347a..0c631bcb86 100644 --- a/modules/audio_coding/codecs/ilbc/enhancer.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h index 661262e42e..d0f5f1a4ed 100644 --- a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/frame_classify.h b/modules/audio_coding/codecs/ilbc/frame_classify.h index 7615106d70..dee67cc5f9 100644 --- a/modules/audio_coding/codecs/ilbc/frame_classify.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/gain_dequant.h b/modules/audio_coding/codecs/ilbc/gain_dequant.h index 2b97550b6c..b5e6cef97b 100644 --- a/modules/audio_coding/codecs/ilbc/gain_dequant.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/gain_quant.h b/modules/audio_coding/codecs/ilbc/gain_quant.h index 761f7d2f79..fab9718a75 100644 --- a/modules/audio_coding/codecs/ilbc/gain_quant.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/modules/audio_coding/codecs/ilbc/get_sync_seq.h index 90962fa063..87030e568f 100644 --- a/modules/audio_coding/codecs/ilbc/get_sync_seq.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/modules/audio_coding/codecs/ilbc/index_conv_dec.h index 4f08ce04df..4d3f733355 100644 --- a/modules/audio_coding/codecs/ilbc/index_conv_dec.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/modules/audio_coding/codecs/ilbc/index_conv_enc.h index 4fbf98084e..0172ac416b 100644 --- a/modules/audio_coding/codecs/ilbc/index_conv_enc.h +++ b/modules/audio_coding/codecs/ilbc/index_conv_enc.h @@ -26,6 +26,6 @@ *---------------------------------------------------------------*/ void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */ - ); +); #endif diff --git a/modules/audio_coding/codecs/ilbc/init_decode.h b/modules/audio_coding/codecs/ilbc/init_decode.h index a2b7b91287..92f9ad68e7 100644 --- a/modules/audio_coding/codecs/ilbc/init_decode.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/init_encode.h b/modules/audio_coding/codecs/ilbc/init_encode.h index 4ada6a30c8..4a233fb946 100644 --- a/modules/audio_coding/codecs/ilbc/init_encode.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/modules/audio_coding/codecs/ilbc/interpolate_samples.h index bc665d7854..f4fa97d477 100644 --- a/modules/audio_coding/codecs/ilbc/interpolate_samples.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lpc_encode.h b/modules/audio_coding/codecs/ilbc/lpc_encode.h index a67b77acbf..ca050b02cc 100644 --- a/modules/audio_coding/codecs/ilbc/lpc_encode.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h index 6cc9d9746d..a0ccfa96ac 100644 --- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h index b278a10f4b..08d1e8325a 100644 --- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h index 6bc6c44dbd..fccc3c2b1c 100644 --- a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/modules/audio_coding/codecs/ilbc/lsf_to_poly.h index f26d3a8d2d..06f292f038 100644 --- a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h +++ b/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 diff --git a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h index c2f4b7692d..a0dfb8e8eb 100644 --- a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h +++ b/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] - #include #include "api/array_view.h" diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc index 795a996624..8161931a7a 100644 --- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc +++ b/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: diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h index 9d2fcfe22e..4b880fb633 100644 --- a/modules/audio_coding/include/audio_coding_module_typedefs.h +++ b/modules/audio_coding/include/audio_coding_module_typedefs.h @@ -64,7 +64,7 @@ struct AudioDecodingCallStats { int calls_to_silence_generator; // Number of calls where silence generated, // and NetEq was disengaged from decoding. int calls_to_neteq; // Number of calls to NetEq. - int decoded_normal; // Number of calls where audio RTP packet decoded. + int decoded_normal; // Number of calls where audio RTP packet decoded. int decoded_neteq_plc; // Number of calls resulted in NetEq PLC. int decoded_codec_plc; // Number of calls resulted in codec PLC. int decoded_cng; // Number of calls where comfort noise generated due to DTX. diff --git a/modules/audio_coding/neteq/accelerate.cc b/modules/audio_coding/neteq/accelerate.cc index f4ef6cdccb..06a38cc534 100644 --- a/modules/audio_coding/neteq/accelerate.cc +++ b/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" diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc index 14ae94649b..3aa49e5b9a 100644 --- a/modules/audio_coding/neteq/audio_multi_vector.cc +++ b/modules/audio_coding/neteq/audio_multi_vector.cc @@ -10,7 +10,6 @@ #include "modules/audio_coding/neteq/audio_multi_vector.h" - #include #include "rtc_base/checks.h" diff --git a/modules/audio_coding/neteq/audio_vector.cc b/modules/audio_coding/neteq/audio_vector.cc index 10e8936447..89105949d5 100644 --- a/modules/audio_coding/neteq/audio_vector.cc +++ b/modules/audio_coding/neteq/audio_vector.cc @@ -10,7 +10,6 @@ #include "modules/audio_coding/neteq/audio_vector.h" - #include #include diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc index 2c42d0d13f..948545d948 100644 --- a/modules/audio_coding/neteq/buffer_level_filter.cc +++ b/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(buffer_size_samples); // Account for time-scale operations (accelerate and pre-emptive expand) and diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc index a2ce888f45..8a906593d7 100644 --- a/modules/audio_coding/neteq/comfort_noise.cc +++ b/modules/audio_coding/neteq/comfort_noise.cc @@ -10,7 +10,6 @@ #include "modules/audio_coding/neteq/comfort_noise.h" - #include #include diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h index 2e64583ec2..bd58788ae0 100644 --- a/modules/audio_coding/neteq/expand.h +++ b/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 #include "modules/audio_coding/neteq/audio_vector.h" diff --git a/modules/audio_coding/neteq/histogram.cc b/modules/audio_coding/neteq/histogram.cc index e4b7f10379..4360d1a904 100644 --- a/modules/audio_coding/neteq/histogram.cc +++ b/modules/audio_coding/neteq/histogram.cc @@ -114,8 +114,8 @@ int Histogram::Quantile(int probability) { // `iat_index`, it is more efficient to start with `sum` = 1 and subtract // elements from the start of the histogram. int inverse_probability = (1 << 30) - probability; - size_t index = 0; // Start from the beginning of `buckets_`. - int sum = 1 << 30; // Assign to 1 in Q30. + size_t index = 0; // Start from the beginning of `buckets_`. + int sum = 1 << 30; // Assign to 1 in Q30. sum -= buckets_[index]; while ((sum > inverse_probability) && (index < buckets_.size() - 1)) { diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc index 6d68fab7fd..f631937535 100644 --- a/modules/audio_coding/neteq/neteq_impl_unittest.cc +++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc @@ -393,7 +393,7 @@ TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) { const int kPayloadLengthSamples = 80; const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit. - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. uint8_t payload[kPayloadLengthBytes] = {0}; RTPHeader rtp_header; rtp_header.payloadType = kPayloadType; @@ -440,7 +440,7 @@ TEST_F(NetEqImplTest, TestDtmfPacketAVT48kHz) { // This test verifies that timestamps propagate from the incoming packets // through to the sync buffer and to the playout timestamp. TEST_F(NetEqImplTest, VerifyTimestampPropagation) { - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const size_t kPayloadLengthSamples = static_cast(10 * kSampleRateHz / 1000); // 10 ms. @@ -559,7 +559,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) { CreateInstance( rtc::make_ref_counted(&mock_decoder)); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const size_t kPayloadLengthSamples = static_cast(10 * kSampleRateHz / 1000); // 10 ms. @@ -674,7 +674,7 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) { UseNoMocks(); CreateInstance(); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const size_t kPayloadLengthSamples = static_cast(10 * kSampleRateHz / 1000); // 10 ms. @@ -767,7 +767,7 @@ TEST_P(NetEqImplTestSampleRateParameter, UseNoMocks(); CreateInstance(); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kPayloadSampleRateHz = 16000; const size_t kPayloadLengthSamples = static_cast(10 * kPayloadSampleRateHz / 1000); // 10 ms. @@ -1004,7 +1004,7 @@ TEST_F(NetEqImplTest, CodecInternalCng) { CreateInstance( rtc::make_ref_counted(&mock_decoder)); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateKhz = 48; const size_t kPayloadLengthSamples = static_cast(20 * kSampleRateKhz); // 20 ms. @@ -1097,7 +1097,7 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) { static const size_t kNetEqMaxFrameSize = 5760; // 120 ms @ 48 kHz. static const size_t kChannels = 2; - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const size_t kPayloadLengthSamples = @@ -1189,7 +1189,7 @@ TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) { const size_t kPayloadLengthSamples = 80; const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit. - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. uint8_t payload[kPayloadLengthBytes] = {0}; RTPHeader rtp_header; rtp_header.payloadType = kPayloadType; @@ -1222,7 +1222,7 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) { CreateInstance( rtc::make_ref_counted(&mock_decoder)); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const size_t kPayloadLengthSamples = static_cast(10 * kSampleRateHz / 1000); // 10 ms. @@ -1281,7 +1281,7 @@ TEST_F(NetEqImplTest, DecodingError) { CreateInstance( rtc::make_ref_counted(&mock_decoder)); - const uint8_t kPayloadType = 17; // Just an arbitrary number. + const uint8_t kPayloadType = 17; // Just an arbitrary number. const int kSampleRateHz = 8000; const int kDecoderErrorCode = -97; // Any negative number. diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index 5317496a83..77bd5b5035 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/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); diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc index 74f841b55c..b0079645ff 100644 --- a/modules/audio_coding/neteq/packet_buffer_unittest.cc +++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc @@ -134,7 +134,7 @@ TEST(PacketBuffer, InsertPacket) { EXPECT_FALSE(buffer.Empty()); EXPECT_EQ(1u, buffer.NumPacketsInBuffer()); const Packet* next_packet = buffer.PeekNextPacket(); - EXPECT_EQ(packet, *next_packet); // Compare contents. + EXPECT_EQ(packet, *next_packet); // Compare contents. EXPECT_CALL(decoder_database, Die()); // Called when object is deleted. // Do not explicitly flush buffer or delete packet to test that it is deleted diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc index a0ba5414ea..55f9bee272 100644 --- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc +++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc @@ -12,7 +12,6 @@ #include "modules/audio_coding/neteq/red_payload_splitter.h" - #include #include // pair diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc index 125d087157..82dd0ae1c1 100644 --- a/modules/audio_coding/neteq/tools/neteq_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_test.cc @@ -313,30 +313,28 @@ NetEqLifetimeStatistics NetEqTest::LifetimeStats() const { } NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() { - DecoderMap codecs = { - {0, SdpAudioFormat("pcmu", 8000, 1)}, - {8, SdpAudioFormat("pcma", 8000, 1)}, + DecoderMap codecs = {{0, SdpAudioFormat("pcmu", 8000, 1)}, + {8, SdpAudioFormat("pcma", 8000, 1)}, #ifdef WEBRTC_CODEC_ILBC - {102, SdpAudioFormat("ilbc", 8000, 1)}, + {102, SdpAudioFormat("ilbc", 8000, 1)}, #endif #ifdef WEBRTC_CODEC_OPUS - {111, SdpAudioFormat("opus", 48000, 2)}, + {111, SdpAudioFormat("opus", 48000, 2)}, #endif - {93, SdpAudioFormat("l16", 8000, 1)}, - {94, SdpAudioFormat("l16", 16000, 1)}, - {95, SdpAudioFormat("l16", 32000, 1)}, - {96, SdpAudioFormat("l16", 48000, 1)}, - {9, SdpAudioFormat("g722", 8000, 1)}, - {106, SdpAudioFormat("telephone-event", 8000, 1)}, - {114, SdpAudioFormat("telephone-event", 16000, 1)}, - {115, SdpAudioFormat("telephone-event", 32000, 1)}, - {116, SdpAudioFormat("telephone-event", 48000, 1)}, - {117, SdpAudioFormat("red", 8000, 1)}, - {13, SdpAudioFormat("cn", 8000, 1)}, - {98, SdpAudioFormat("cn", 16000, 1)}, - {99, SdpAudioFormat("cn", 32000, 1)}, - {100, SdpAudioFormat("cn", 48000, 1)} - }; + {93, SdpAudioFormat("l16", 8000, 1)}, + {94, SdpAudioFormat("l16", 16000, 1)}, + {95, SdpAudioFormat("l16", 32000, 1)}, + {96, SdpAudioFormat("l16", 48000, 1)}, + {9, SdpAudioFormat("g722", 8000, 1)}, + {106, SdpAudioFormat("telephone-event", 8000, 1)}, + {114, SdpAudioFormat("telephone-event", 16000, 1)}, + {115, SdpAudioFormat("telephone-event", 32000, 1)}, + {116, SdpAudioFormat("telephone-event", 48000, 1)}, + {117, SdpAudioFormat("red", 8000, 1)}, + {13, SdpAudioFormat("cn", 8000, 1)}, + {98, SdpAudioFormat("cn", 16000, 1)}, + {99, SdpAudioFormat("cn", 32000, 1)}, + {100, SdpAudioFormat("cn", 48000, 1)}}; return codecs; } diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h index 96710907df..a4b3da9a4b 100644 --- a/modules/audio_coding/neteq/tools/packet.h +++ b/modules/audio_coding/neteq/tools/packet.h @@ -95,7 +95,7 @@ class Packet { // Virtual lengths are used when parsing RTP header files (dummy RTP files). const size_t virtual_packet_length_bytes_; size_t virtual_payload_length_bytes_ = 0; - const double time_ms_; // Used to denote a packet's arrival time. + const double time_ms_; // Used to denote a packet's arrival time. const bool valid_header_; }; diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc index a43c29638c..7a8daef945 100644 --- a/modules/audio_coding/neteq/tools/rtp_file_source.cc +++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc @@ -80,8 +80,7 @@ std::unique_ptr RtpFileSource::NextPacket() { } RtpFileSource::RtpFileSource(absl::optional ssrc_filter) - : PacketSource(), - ssrc_filter_(ssrc_filter) {} + : PacketSource(), ssrc_filter_(ssrc_filter) {} bool RtpFileSource::OpenFile(absl::string_view file_name) { rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name)); diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc index e883fc11d6..5633f11b86 100644 --- a/modules/audio_coding/neteq/tools/rtp_generator.cc +++ b/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 { diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index 014a1d2d54..cab36458e0 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -229,8 +229,8 @@ EncodeDecodeTest::EncodeDecodeTest() = default; void EncodeDecodeTest::Perform() { const std::map send_codecs = { - {107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}}, - {109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}}, + {107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}}, + {109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}}, {8, {"PCMA", 8000, 1}}, #ifdef WEBRTC_CODEC_ILBC {102, {"ILBC", 8000, 1}}, diff --git a/modules/audio_device/linux/audio_device_alsa_linux.cc b/modules/audio_device/linux/audio_device_alsa_linux.cc index 50cf3beb6c..eab73737c5 100644 --- a/modules/audio_device/linux/audio_device_alsa_linux.cc +++ b/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" @@ -1053,8 +1052,8 @@ int32_t AudioDeviceLinuxALSA::StartRecording() { } int32_t AudioDeviceLinuxALSA::StopRecording() { - MutexLock lock(&mutex_); - return StopRecordingLocked(); + MutexLock lock(&mutex_); + return StopRecordingLocked(); } int32_t AudioDeviceLinuxALSA::StopRecordingLocked() { @@ -1157,8 +1156,8 @@ int32_t AudioDeviceLinuxALSA::StartPlayout() { } int32_t AudioDeviceLinuxALSA::StopPlayout() { - MutexLock lock(&mutex_); - return StopPlayoutLocked(); + MutexLock lock(&mutex_); + return StopPlayoutLocked(); } int32_t AudioDeviceLinuxALSA::StopPlayoutLocked() { diff --git a/modules/audio_device/mac/audio_device_mac.cc b/modules/audio_device/mac/audio_device_mac.cc index 527f76a371..ed7b0e4669 100644 --- a/modules/audio_device/mac/audio_device_mac.cc +++ b/modules/audio_device/mac/audio_device_mac.cc @@ -11,8 +11,8 @@ #include "modules/audio_device/mac/audio_device_mac.h" #include -#include // mach_task_self() -#include // sysctlbyname() +#include // mach_task_self() +#include // sysctlbyname() #include @@ -1355,8 +1355,8 @@ int32_t AudioDeviceMac::StopRecording() { // rendering has ended before stopping itself. if (_recording && captureDeviceIsAlive == 1) { _recording = false; - _doStop = true; // Signal to io proc to stop audio device - mutex_.Unlock(); // Cannot be under lock, risk of deadlock + _doStop = true; // Signal to io proc to stop audio device + mutex_.Unlock(); // Cannot be under lock, risk of deadlock if (!_stopEvent.Wait(TimeDelta::Seconds(2))) { MutexLock lockScoped(&mutex_); RTC_LOG(LS_WARNING) << "Timed out stopping the shared IOProc." @@ -1465,8 +1465,8 @@ int32_t AudioDeviceMac::StopPlayout() { // In the case of a shared device, the IOProc will verify capturing // has ended before stopping itself. _playing = false; - _doStop = true; // Signal to io proc to stop audio device - mutex_.Unlock(); // Cannot be under lock, risk of deadlock + _doStop = true; // Signal to io proc to stop audio device + mutex_.Unlock(); // Cannot be under lock, risk of deadlock if (!_stopEvent.Wait(TimeDelta::Seconds(2))) { MutexLock lockScoped(&mutex_); RTC_LOG(LS_WARNING) << "Timed out stopping the render IOProc." diff --git a/modules/audio_device/win/audio_device_core_win.cc b/modules/audio_device/win/audio_device_core_win.cc index 1e3a94edf6..aa8b6a9ebe 100644 --- a/modules/audio_device/win/audio_device_core_win.cc +++ b/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 - #include #include #include #include +#include #include #include #include @@ -3256,9 +3255,10 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() { QueryPerformanceCounter(&t1); // Get the current recording and playout delay. - uint32_t sndCardRecDelay = (uint32_t)( - ((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) / 10000) + - (10 * syncBufIndex) / _recBlockSize - 10); + uint32_t sndCardRecDelay = + (uint32_t)(((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime) / + 10000) + + (10 * syncBufIndex) / _recBlockSize - 10); uint32_t sndCardPlayDelay = static_cast(_sndCardPlayDelay); while (syncBufIndex >= _recBlockSize) { diff --git a/modules/audio_device/win/audio_device_core_win.h b/modules/audio_device/win/audio_device_core_win.h index 7e7ef21157..380effb449 100644 --- a/modules/audio_device/win/audio_device_core_win.h +++ b/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 // CLSID_CWMAudioAEC +//(must be before audioclient.h) -#include "modules/audio_device/audio_device_generic.h" - -#include // CLSID_CWMAudioAEC - // (must be before audioclient.h) #include // WASAPI #include #include // Avrt @@ -27,8 +24,10 @@ #include // 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") diff --git a/modules/audio_device/win/core_audio_utility_win_unittest.cc b/modules/audio_device/win/core_audio_utility_win_unittest.cc index 277f54eb35..fc4a610eef 100644 --- a/modules/audio_device/win/core_audio_utility_win_unittest.cc +++ b/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" diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc index 6c8c948026..b6eda9f117 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc +++ b/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 +#include "modules/audio_processing/aec3/adaptive_fir_filter.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc index 5fe7514db1..1e63cf8fe7 100644 --- a/modules/audio_processing/aec3/adaptive_fir_filter_erl_avx2.cc +++ b/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 +#include "modules/audio_processing/aec3/adaptive_fir_filter_erl.h" + namespace webrtc { namespace aec3 { diff --git a/modules/audio_processing/aec3/echo_path_variability.h b/modules/audio_processing/aec3/echo_path_variability.h index 78e4f64b2b..45b33c217c 100644 --- a/modules/audio_processing/aec3/echo_path_variability.h +++ b/modules/audio_processing/aec3/echo_path_variability.h @@ -14,11 +14,7 @@ namespace webrtc { struct EchoPathVariability { - enum class DelayAdjustment { - kNone, - kBufferFlush, - kNewDetectedDelay - }; + enum class DelayAdjustment { kNone, kBufferFlush, kNewDetectedDelay }; EchoPathVariability(bool gain_change, DelayAdjustment delay_change, diff --git a/modules/audio_processing/aec3/fft_data_avx2.cc b/modules/audio_processing/aec3/fft_data_avx2.cc index 1fe4bd69c6..a4b3056071 100644 --- a/modules/audio_processing/aec3/fft_data_avx2.cc +++ b/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 #include "api/array_view.h" +#include "modules/audio_processing/aec3/fft_data.h" namespace webrtc { diff --git a/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc b/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc index 8d38dd0991..3b6e942d88 100644 --- a/modules/audio_processing/aec3/multi_channel_content_detector_unittest.cc +++ b/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( - 1, metrics::NumEvents("WebRTC.Audio.EchoCanceller." - "ProcessingPersistentMultichannelContent", 0)); - EXPECT_METRIC_EQ( - 1, metrics::NumEvents("WebRTC.Audio.EchoCanceller." - "ProcessingPersistentMultichannelContent", 1)); + EXPECT_METRIC_EQ(1, + metrics::NumEvents("WebRTC.Audio.EchoCanceller." + "ProcessingPersistentMultichannelContent", + 0)); + EXPECT_METRIC_EQ(1, + metrics::NumEvents("WebRTC.Audio.EchoCanceller." + "ProcessingPersistentMultichannelContent", + 1)); } } // namespace webrtc diff --git a/modules/audio_processing/aec3/reverb_model.h b/modules/audio_processing/aec3/reverb_model.h index 5ba54853da..47ed2f78f3 100644 --- a/modules/audio_processing/aec3/reverb_model.h +++ b/modules/audio_processing/aec3/reverb_model.h @@ -49,7 +49,6 @@ class ReverbModel { float reverb_decay); private: - std::array reverb_; }; diff --git a/modules/audio_processing/aec3/vector_math_avx2.cc b/modules/audio_processing/aec3/vector_math_avx2.cc index 0b5f3c142e..a9805daf88 100644 --- a/modules/audio_processing/aec3/vector_math_avx2.cc +++ b/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 #include #include "api/array_view.h" +#include "modules/audio_processing/aec3/vector_math.h" #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/audio_processing/aecm/aecm_core.cc b/modules/audio_processing/aecm/aecm_core.cc index fbc3239732..b4631460ca 100644 --- a/modules/audio_processing/aecm/aecm_core.cc +++ b/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. // diff --git a/modules/audio_processing/aecm/aecm_core_c.cc b/modules/audio_processing/aecm/aecm_core_c.cc index d363dd2cfd..59e0296bbf 100644 --- a/modules/audio_processing/aecm/aecm_core_c.cc +++ b/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)( - (scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14); + fft[PART_LEN + i] = (int16_t)((scaled_time_signal * + 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)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14)); - efw[i].imag = (int16_t)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14)); + efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, + hnl[i], 14)); + efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, + hnl[i], 14)); } } else { // multiply with Wiener coefficients for (i = 0; i < PART_LEN1; i++) { - efw[i].real = (int16_t)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14)); - efw[i].imag = (int16_t)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14)); + efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, + hnl[i], 14)); + efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, + hnl[i], 14)); } } diff --git a/modules/audio_processing/aecm/aecm_core_mips.cc b/modules/audio_processing/aecm/aecm_core_mips.cc index 828aa6d2fb..16b03cfe51 100644 --- a/modules/audio_processing/aecm/aecm_core_mips.cc +++ b/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), - [WebRtcAecm_kSqrtHanning] "r"(WebRtcAecm_kSqrtHanning) + : [out_aecm] "r"(out_aecm), [WebRtcAecm_kSqrtHanning] "r"( + 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)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14)); - efw[i].imag = (int16_t)( - WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14)); + efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, + hnl[i], 14)); + efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, + 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"(minTrackShift)); + : [outLShift32] "r"(outLShift32), [minTrackShift] "r"( + 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"(minTrackShift)); + : [outLShift32] "r"(outLShift32), [minTrackShift] "r"( + minTrackShift)); } } else { // Reset "too high" counter diff --git a/modules/audio_processing/agc/legacy/analog_agc.h b/modules/audio_processing/agc/legacy/analog_agc.h index 22cd924a93..7a231c8a64 100644 --- a/modules/audio_processing/agc/legacy/analog_agc.h +++ b/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" @@ -63,7 +62,7 @@ typedef struct { int32_t upperSecondaryLimit; // = kRxxBufferLen * 2677832; -17 dBfs int32_t lowerSecondaryLimit; // = kRxxBufferLen * 267783; -27 dBfs uint16_t targetIdx; // Table index for corresponding target level - int16_t analogTarget; // Digital reference level in ENV scale + int16_t analogTarget; // Digital reference level in ENV scale // Analog AGC specific variables int32_t filterState[8]; // For downsampling wb to nb @@ -74,8 +73,8 @@ typedef struct { int32_t Rxx160_LPw32; // Low pass filtered frame energies int32_t Rxx16_LPw32Max; // Keeps track of largest energy subframe int32_t Rxx16_vectorw32[kRxxBufferLen]; // Array with subframe energies - int32_t Rxx16w32_array[2][5]; // Energy values of microphone signal - int32_t env[2][10]; // Envelope values of subframes + int32_t Rxx16w32_array[2][5]; // Energy values of microphone signal + int32_t env[2][10]; // Envelope values of subframes int16_t Rxx16pos; // Current position in the Rxx16_vectorw32 int16_t envSum; // Filtered scaled envelope in subframes diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc b/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc index 91501fb6e3..a13e77461a 100644 --- a/modules/audio_processing/agc2/rnn_vad/rnn_fc.cc +++ b/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 #include -#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" diff --git a/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc b/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc index e4d246d9ab..a875e11daf 100644 --- a/modules/audio_processing/agc2/rnn_vad/vector_math_avx2.cc +++ b/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 #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" diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc index edc49d1401..5f2b4872b9 100644 --- a/modules/audio_processing/gain_control_impl.cc +++ b/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; diff --git a/modules/audio_processing/ns/noise_suppressor.cc b/modules/audio_processing/ns/noise_suppressor.cc index d66faa6ed4..7c524dadf3 100644 --- a/modules/audio_processing/ns/noise_suppressor.cc +++ b/modules/audio_processing/ns/noise_suppressor.cc @@ -13,6 +13,7 @@ #include #include #include + #include #include "modules/audio_processing/ns/fast_math.h" diff --git a/modules/audio_processing/ns/prior_signal_model_estimator.cc b/modules/audio_processing/ns/prior_signal_model_estimator.cc index c814658e57..f77dcd6dac 100644 --- a/modules/audio_processing/ns/prior_signal_model_estimator.cc +++ b/modules/audio_processing/ns/prior_signal_model_estimator.cc @@ -11,6 +11,7 @@ #include "modules/audio_processing/ns/prior_signal_model_estimator.h" #include + #include #include "modules/audio_processing/ns/fast_math.h" diff --git a/modules/audio_processing/ns/quantile_noise_estimator.h b/modules/audio_processing/ns/quantile_noise_estimator.h index 67d1512209..55b0bfa3fe 100644 --- a/modules/audio_processing/ns/quantile_noise_estimator.h +++ b/modules/audio_processing/ns/quantile_noise_estimator.h @@ -12,6 +12,7 @@ #define MODULES_AUDIO_PROCESSING_NS_QUANTILE_NOISE_ESTIMATOR_H_ #include + #include #include "api/array_view.h" diff --git a/modules/audio_processing/ns/speech_probability_estimator.cc b/modules/audio_processing/ns/speech_probability_estimator.cc index fce9bc8e07..65f17f4af2 100644 --- a/modules/audio_processing/ns/speech_probability_estimator.cc +++ b/modules/audio_processing/ns/speech_probability_estimator.cc @@ -11,6 +11,7 @@ #include "modules/audio_processing/ns/speech_probability_estimator.h" #include + #include #include "modules/audio_processing/ns/fast_math.h" diff --git a/modules/audio_processing/ns/wiener_filter.cc b/modules/audio_processing/ns/wiener_filter.cc index e14b7970d9..1eb50a7166 100644 --- a/modules/audio_processing/ns/wiener_filter.cc +++ b/modules/audio_processing/ns/wiener_filter.cc @@ -13,6 +13,7 @@ #include #include #include + #include #include "modules/audio_processing/ns/fast_math.h" diff --git a/modules/audio_processing/test/conversational_speech/generator.cc b/modules/audio_processing/test/conversational_speech/generator.cc index d0bc2f2319..4f776fa216 100644 --- a/modules/audio_processing/test/conversational_speech/generator.cc +++ b/modules/audio_processing/test/conversational_speech/generator.cc @@ -9,9 +9,8 @@ */ #include -#include - #include +#include #include "absl/flags/flag.h" #include "absl/flags/parse.h" diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc index b44f2e7d21..0483dbf54d 100644 --- a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc +++ b/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) { - c->loss_rate = 0.15; - }); + net->UpdateConfig([](NetworkSimulationConfig* c) { c->loss_rate = 0.15; }); s.RunFor(TimeDelta::Seconds(20)); // Bandwidth decreases thanks to loss based bwe v0. diff --git a/modules/congestion_controller/pcc/bitrate_controller.cc b/modules/congestion_controller/pcc/bitrate_controller.cc index 16b8e6966f..1a9cddb519 100644 --- a/modules/congestion_controller/pcc/bitrate_controller.cc +++ b/modules/congestion_controller/pcc/bitrate_controller.cc @@ -17,7 +17,6 @@ #include #include - namespace webrtc { namespace pcc { diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/modules/congestion_controller/rtp/transport_feedback_adapter.cc index e83d09d263..be17e50472 100644 --- a/modules/congestion_controller/rtp/transport_feedback_adapter.cc +++ b/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) { @@ -213,51 +212,52 @@ TransportFeedbackAdapter::ProcessTransportFeedbackInner( size_t failed_lookups = 0; size_t ignored = 0; - feedback.ForAllPackets([&](uint16_t sequence_number, - TimeDelta delta_since_base) { - int64_t seq_num = seq_num_unwrapper_.Unwrap(sequence_number); + feedback.ForAllPackets( + [&](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 - // 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); - } - last_ack_seq_num_ = seq_num; - } + if (seq_num > last_ack_seq_num_) { + // Starts at history_.begin() if last_ack_seq_num_ < 0, since any + // 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); + } + last_ack_seq_num_ = seq_num; + } - auto it = history_.find(seq_num); - if (it == history_.end()) { - ++failed_lookups; - return; - } + auto it = history_.find(seq_num); + if (it == history_.end()) { + ++failed_lookups; + return; + } - if (it->second.sent.send_time.IsInfinite()) { - // TODO(srte): Fix the tests that makes this happen and make this a - // DCHECK. - RTC_DLOG(LS_ERROR) - << "Received feedback before packet was indicated as sent"; - return; - } + if (it->second.sent.send_time.IsInfinite()) { + // TODO(srte): Fix the tests that makes this happen and make this a + // DCHECK. + RTC_DLOG(LS_ERROR) + << "Received feedback before packet was indicated as sent"; + return; + } - PacketFeedback packet_feedback = it->second; - if (delta_since_base.IsFinite()) { - packet_feedback.receive_time = - current_offset_ + delta_since_base.RoundDownTo(TimeDelta::Millis(1)); - // 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_) { - PacketResult result; - result.sent_packet = packet_feedback.sent; - result.receive_time = packet_feedback.receive_time; - packet_result_vector.push_back(result); - } else { - ++ignored; - } - }); + PacketFeedback packet_feedback = it->second; + if (delta_since_base.IsFinite()) { + packet_feedback.receive_time = + current_offset_ + + delta_since_base.RoundDownTo(TimeDelta::Millis(1)); + // 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_) { + PacketResult result; + result.sent_packet = packet_feedback.sent; + result.receive_time = packet_feedback.receive_time; + packet_result_vector.push_back(result); + } else { + ++ignored; + } + }); if (failed_lookups > 0) { RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer.cc index 469c21434a..5a6a2e1e9b 100644 --- a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc +++ b/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" diff --git a/modules/desktop_capture/desktop_frame_unittest.cc b/modules/desktop_capture/desktop_frame_unittest.cc index 22df1a7f53..e690e6ae5b 100644 --- a/modules/desktop_capture/desktop_frame_unittest.cc +++ b/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( - *src_frame, test.horizontal_scale, test.vertical_scale); + dest_frame->CopyIntersectingPixelsFrom(*src_frame, test.horizontal_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)); } diff --git a/modules/desktop_capture/full_screen_application_handler.cc b/modules/desktop_capture/full_screen_application_handler.cc index e0975570ba..68ee8321b4 100644 --- a/modules/desktop_capture/full_screen_application_handler.cc +++ b/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 { diff --git a/modules/desktop_capture/full_screen_window_detector.cc b/modules/desktop_capture/full_screen_window_detector.cc index d0bc9c7ca6..956a0b4663 100644 --- a/modules/desktop_capture/full_screen_window_detector.cc +++ b/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" diff --git a/modules/desktop_capture/linux/wayland/screencast_portal.cc b/modules/desktop_capture/linux/wayland/screencast_portal.cc index 52788524c8..3d28d42ba6 100644 --- a/modules/desktop_capture/linux/wayland/screencast_portal.cc +++ b/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 diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc index 21863b7d8d..1eea8bfbf5 100644 --- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc +++ b/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_context_, fcntl(fd, F_DUPFD_CLOEXEC), nullptr, 0); + pw_core_ = pw_context_connect_fd(pw_context_, fcntl(fd, F_DUPFD_CLOEXEC), + nullptr, 0); } else { pw_core_ = pw_context_connect(pw_context_, nullptr, 0); } diff --git a/modules/desktop_capture/linux/x11/x_error_trap.cc b/modules/desktop_capture/linux/x11/x_error_trap.cc index 94e3a03c73..24c2065111 100644 --- a/modules/desktop_capture/linux/x11/x_error_trap.cc +++ b/modules/desktop_capture/linux/x11/x_error_trap.cc @@ -10,10 +10,10 @@ #include "modules/desktop_capture/linux/x11/x_error_trap.h" -#include - #include +#include + #include "rtc_base/checks.h" namespace webrtc { diff --git a/modules/desktop_capture/mac/full_screen_mac_application_handler.h b/modules/desktop_capture/mac/full_screen_mac_application_handler.h index f795a22030..060cdb5a61 100644 --- a/modules/desktop_capture/mac/full_screen_mac_application_handler.h +++ b/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 + #include "modules/desktop_capture/full_screen_application_handler.h" namespace webrtc { diff --git a/modules/desktop_capture/mac/screen_capturer_mac.h b/modules/desktop_capture/mac/screen_capturer_mac.h index d9a5966efa..7e38b5bd08 100644 --- a/modules/desktop_capture/mac/screen_capturer_mac.h +++ b/modules/desktop_capture/mac/screen_capturer_mac.h @@ -36,10 +36,9 @@ class DisplayStreamManager; // A class to perform video frame capturing for mac. class ScreenCapturerMac final : public DesktopCapturer { public: - ScreenCapturerMac( - rtc::scoped_refptr desktop_config_monitor, - bool detect_updated_region, - bool allow_iosurface); + ScreenCapturerMac(rtc::scoped_refptr desktop_config_monitor, + bool detect_updated_region, + bool allow_iosurface); ~ScreenCapturerMac() override; ScreenCapturerMac(const ScreenCapturerMac&) = delete; diff --git a/modules/desktop_capture/mac/window_list_utils.h b/modules/desktop_capture/mac/window_list_utils.h index a9b1e7007c..34d1313234 100644 --- a/modules/desktop_capture/mac/window_list_utils.h +++ b/modules/desktop_capture/mac/window_list_utils.h @@ -14,6 +14,7 @@ #include #include + #include "api/function_view.h" #include "modules/desktop_capture/desktop_capture_types.h" #include "modules/desktop_capture/desktop_capturer.h" diff --git a/modules/desktop_capture/screen_capturer_helper.cc b/modules/desktop_capture/screen_capturer_helper.cc index f8261a90b0..04e72b3e9f 100644 --- a/modules/desktop_capture/screen_capturer_helper.cc +++ b/modules/desktop_capture/screen_capturer_helper.cc @@ -10,7 +10,6 @@ #include "modules/desktop_capture/screen_capturer_helper.h" - namespace webrtc { void ScreenCapturerHelper::ClearInvalidRegion() { diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.h b/modules/desktop_capture/win/full_screen_win_application_handler.h index c97cbe252b..286e8e5cb9 100644 --- a/modules/desktop_capture/win/full_screen_win_application_handler.h +++ b/modules/desktop_capture/win/full_screen_win_application_handler.h @@ -12,6 +12,7 @@ #define MODULES_DESKTOP_CAPTURE_WIN_FULL_SCREEN_WIN_APPLICATION_HANDLER_H_ #include + #include "modules/desktop_capture/full_screen_application_handler.h" namespace webrtc { diff --git a/modules/desktop_capture/win/screen_capture_utils.h b/modules/desktop_capture/win/screen_capture_utils.h index 9aa838ab8d..71c79b9ab3 100644 --- a/modules/desktop_capture/win/screen_capture_utils.h +++ b/modules/desktop_capture/win/screen_capture_utils.h @@ -15,8 +15,8 @@ // Forward declare HMONITOR in a windows.h compatible way so that we can avoid // including windows.h. #define WEBRTC_DECLARE_HANDLE(name) \ -struct name##__; \ -typedef struct name##__* name + struct name##__; \ + typedef struct name##__* name WEBRTC_DECLARE_HANDLE(HMONITOR); #undef WEBRTC_DECLARE_HANDLE #endif diff --git a/modules/pacing/pacing_controller_unittest.cc b/modules/pacing/pacing_controller_unittest.cc index d2ee908691..2c5f8e9b07 100644 --- a/modules/pacing/pacing_controller_unittest.cc +++ b/modules/pacing/pacing_controller_unittest.cc @@ -656,55 +656,55 @@ TEST_F(PacingControllerTest, Padding) { AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); } - const TimeDelta actual_pace_time = clock_.CurrentTime() - start_time; - EXPECT_LE((actual_pace_time - expected_pace_time).Abs(), - PacingController::kMinSleepTime); + const TimeDelta actual_pace_time = clock_.CurrentTime() - start_time; + EXPECT_LE((actual_pace_time - expected_pace_time).Abs(), + PacingController::kMinSleepTime); - // Pacing media happens at 2.5x, but padding was configured with 1.0x - // factor. We have to wait until the padding debt is gone before we start - // sending padding. - const TimeDelta time_to_padding_debt_free = - (expected_pace_time * kPaceMultiplier) - actual_pace_time; - clock_.AdvanceTime(time_to_padding_debt_free - - PacingController::kMinSleepTime); + // Pacing media happens at 2.5x, but padding was configured with 1.0x + // factor. We have to wait until the padding debt is gone before we start + // sending padding. + const TimeDelta time_to_padding_debt_free = + (expected_pace_time * kPaceMultiplier) - actual_pace_time; + clock_.AdvanceTime(time_to_padding_debt_free - + PacingController::kMinSleepTime); + pacer->ProcessPackets(); + + // Send 10 padding packets. + const size_t kPaddingPacketsToSend = 10; + DataSize padding_sent = DataSize::Zero(); + size_t packets_sent = 0; + Timestamp first_send_time = Timestamp::MinusInfinity(); + Timestamp last_send_time = Timestamp::MinusInfinity(); + + EXPECT_CALL(callback_, SendPadding) + .Times(kPaddingPacketsToSend) + .WillRepeatedly([&](size_t target_size) { + ++packets_sent; + if (packets_sent < kPaddingPacketsToSend) { + // Don't count bytes of last packet, instead just + // use this as the time the last packet finished + // sending. + padding_sent += DataSize::Bytes(target_size); + } + if (first_send_time.IsInfinite()) { + first_send_time = clock_.CurrentTime(); + } else { + last_send_time = clock_.CurrentTime(); + } + return target_size; + }); + EXPECT_CALL(callback_, SendPacket(_, _, _, false, true)) + .Times(kPaddingPacketsToSend); + + while (packets_sent < kPaddingPacketsToSend) { + AdvanceTimeUntil(pacer->NextSendTime()); pacer->ProcessPackets(); + } - // Send 10 padding packets. - const size_t kPaddingPacketsToSend = 10; - DataSize padding_sent = DataSize::Zero(); - size_t packets_sent = 0; - Timestamp first_send_time = Timestamp::MinusInfinity(); - Timestamp last_send_time = Timestamp::MinusInfinity(); - - EXPECT_CALL(callback_, SendPadding) - .Times(kPaddingPacketsToSend) - .WillRepeatedly([&](size_t target_size) { - ++packets_sent; - if (packets_sent < kPaddingPacketsToSend) { - // Don't count bytes of last packet, instead just - // use this as the time the last packet finished - // sending. - padding_sent += DataSize::Bytes(target_size); - } - if (first_send_time.IsInfinite()) { - first_send_time = clock_.CurrentTime(); - } else { - last_send_time = clock_.CurrentTime(); - } - return target_size; - }); - EXPECT_CALL(callback_, SendPacket(_, _, _, false, true)) - .Times(kPaddingPacketsToSend); - - while (packets_sent < kPaddingPacketsToSend) { - AdvanceTimeUntil(pacer->NextSendTime()); - pacer->ProcessPackets(); - } - - // Verify rate of sent padding. - TimeDelta padding_duration = last_send_time - first_send_time; - DataRate padding_rate = padding_sent / padding_duration; - EXPECT_EQ(padding_rate, kTargetRate); + // Verify rate of sent padding. + TimeDelta padding_duration = last_send_time - first_send_time; + DataRate padding_rate = padding_sent / padding_duration; + EXPECT_EQ(padding_rate, kTargetRate); } TEST_F(PacingControllerTest, NoPaddingBeforeNormalPacket) { @@ -1245,66 +1245,66 @@ TEST_F(PacingControllerTest, SkipsProbesWhenProcessIntervalTooLarge) { pacer->ProcessPackets(); } - // Probe at a very high rate. - std::vector probe_clusters = { - {.at_time = clock_.CurrentTime(), - .target_data_rate = DataRate::KilobitsPerSec(10000), // 10 Mbps, - .target_duration = TimeDelta::Millis(15), - .target_probe_count = 5, - .id = kProbeClusterId}}; - pacer->CreateProbeClusters(probe_clusters); + // Probe at a very high rate. + std::vector probe_clusters = { + {.at_time = clock_.CurrentTime(), + .target_data_rate = DataRate::KilobitsPerSec(10000), // 10 Mbps, + .target_duration = TimeDelta::Millis(15), + .target_probe_count = 5, + .id = kProbeClusterId}}; + pacer->CreateProbeClusters(probe_clusters); - // We need one packet to start the probe. - pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, - sequence_number++, - clock_.TimeInMilliseconds(), kPacketSize)); - const int packets_sent_before_probe = packet_sender.packets_sent(); - AdvanceTimeUntil(pacer->NextSendTime()); - pacer->ProcessPackets(); - EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 1); + // We need one packet to start the probe. + pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, + sequence_number++, + clock_.TimeInMilliseconds(), kPacketSize)); + const int packets_sent_before_probe = packet_sender.packets_sent(); + AdvanceTimeUntil(pacer->NextSendTime()); + pacer->ProcessPackets(); + EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 1); - // Figure out how long between probe packets. - Timestamp start_time = clock_.CurrentTime(); - AdvanceTimeUntil(pacer->NextSendTime()); - TimeDelta time_between_probes = clock_.CurrentTime() - start_time; - // Advance that distance again + 1ms. - clock_.AdvanceTime(time_between_probes); + // Figure out how long between probe packets. + Timestamp start_time = clock_.CurrentTime(); + AdvanceTimeUntil(pacer->NextSendTime()); + TimeDelta time_between_probes = clock_.CurrentTime() - start_time; + // Advance that distance again + 1ms. + clock_.AdvanceTime(time_between_probes); - // Send second probe packet. - pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, - sequence_number++, - clock_.TimeInMilliseconds(), kPacketSize)); - pacer->ProcessPackets(); - EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 2); - PacedPacketInfo last_pacing_info = packet_sender.last_pacing_info(); - EXPECT_EQ(last_pacing_info.probe_cluster_id, kProbeClusterId); + // Send second probe packet. + pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, ssrc, + sequence_number++, + clock_.TimeInMilliseconds(), kPacketSize)); + pacer->ProcessPackets(); + EXPECT_EQ(packet_sender.packets_sent(), packets_sent_before_probe + 2); + PacedPacketInfo last_pacing_info = packet_sender.last_pacing_info(); + EXPECT_EQ(last_pacing_info.probe_cluster_id, kProbeClusterId); - // We're exactly where we should be for the next probe. - const Timestamp probe_time = clock_.CurrentTime(); - EXPECT_EQ(pacer->NextSendTime(), clock_.CurrentTime()); + // We're exactly where we should be for the next probe. + const Timestamp probe_time = clock_.CurrentTime(); + EXPECT_EQ(pacer->NextSendTime(), clock_.CurrentTime()); - BitrateProberConfig probing_config(&trials); - EXPECT_GT(probing_config.max_probe_delay.Get(), TimeDelta::Zero()); - // Advance to within max probe delay, should still return same target. - clock_.AdvanceTime(probing_config.max_probe_delay.Get()); - EXPECT_EQ(pacer->NextSendTime(), probe_time); + BitrateProberConfig probing_config(&trials); + EXPECT_GT(probing_config.max_probe_delay.Get(), TimeDelta::Zero()); + // Advance to within max probe delay, should still return same target. + clock_.AdvanceTime(probing_config.max_probe_delay.Get()); + EXPECT_EQ(pacer->NextSendTime(), probe_time); - // Too high probe delay, drop it! - clock_.AdvanceTime(TimeDelta::Micros(1)); + // Too high probe delay, drop it! + clock_.AdvanceTime(TimeDelta::Micros(1)); - int packets_sent_before_timeout = packet_sender.total_packets_sent(); - // Expected next process time is unchanged, but calling should not - // generate new packets. - EXPECT_EQ(pacer->NextSendTime(), probe_time); - pacer->ProcessPackets(); - EXPECT_EQ(packet_sender.total_packets_sent(), packets_sent_before_timeout); + int packets_sent_before_timeout = packet_sender.total_packets_sent(); + // Expected next process time is unchanged, but calling should not + // generate new packets. + EXPECT_EQ(pacer->NextSendTime(), probe_time); + pacer->ProcessPackets(); + EXPECT_EQ(packet_sender.total_packets_sent(), packets_sent_before_timeout); - // Next packet sent is not part of probe. - AdvanceTimeUntil(pacer->NextSendTime()); - pacer->ProcessPackets(); - const int expected_probe_id = PacedPacketInfo::kNotAProbe; - EXPECT_EQ(packet_sender.last_pacing_info().probe_cluster_id, - expected_probe_id); + // Next packet sent is not part of probe. + AdvanceTimeUntil(pacer->NextSendTime()); + pacer->ProcessPackets(); + const int expected_probe_id = PacedPacketInfo::kNotAProbe; + EXPECT_EQ(packet_sender.last_pacing_info().probe_cluster_id, + expected_probe_id); } TEST_F(PacingControllerTest, ProbingWithPaddingSupport) { diff --git a/modules/pacing/packet_router_unittest.cc b/modules/pacing/packet_router_unittest.cc index 65b2ad24d6..76d14c9dce 100644 --- a/modules/pacing/packet_router_unittest.cc +++ b/modules/pacing/packet_router_unittest.cc @@ -75,7 +75,6 @@ TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_GeneratePadding) { EXPECT_TRUE(packet_router_.GeneratePadding(bytes).empty()); } - TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendRemb) { const std::vector ssrcs = {1, 2, 3}; constexpr uint32_t bitrate_bps = 10000; diff --git a/modules/remote_bitrate_estimator/aimd_rate_control.cc b/modules/remote_bitrate_estimator/aimd_rate_control.cc index 129e82492f..b9abb785a5 100644 --- a/modules/remote_bitrate_estimator/aimd_rate_control.cc +++ b/modules/remote_bitrate_estimator/aimd_rate_control.cc @@ -217,7 +217,7 @@ double AimdRateControl::GetNearMaxIncreaseRateBpsPerSecond() const { // Approximate the over-use estimator delay to 100 ms. TimeDelta response_time = rtt_ + TimeDelta::Millis(100); - response_time = response_time * 2; + response_time = response_time * 2; double increase_rate_bps_per_second = (avg_packet_size / response_time).bps(); double kMinIncreaseRateBpsPerSecond = 4000; diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc index 024dc23df1..20326286fa 100644 --- a/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc +++ b/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc @@ -10,7 +10,6 @@ #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" - #include #include diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc index 1dc756d876..7c6c2cf443 100644 --- a/modules/rtp_rtcp/source/receive_statistics_impl.cc +++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc @@ -33,7 +33,8 @@ constexpr int64_t kStatisticsProcessIntervalMs = 1000; StreamStatistician::~StreamStatistician() {} -StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock, +StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, + Clock* clock, int max_reordering_threshold) : ssrc_(ssrc), clock_(clock), diff --git a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc index 79d57bc991..979609ab40 100644 --- a/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc +++ b/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc @@ -549,15 +549,14 @@ TEST(RtcpReceiverTest, EXPECT_THAT( receiver.GetLatestReportBlockData(), - UnorderedElementsAre( - Property( - &ReportBlockData::report_block, - AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverMainSsrc), - Field(&RTCPReportBlock::sender_ssrc, kSenderSsrc2), - Field(&RTCPReportBlock::fraction_lost, kFracLost[1]), - Field(&RTCPReportBlock::packets_lost, kCumLost[1]), - Field(&RTCPReportBlock::extended_highest_sequence_number, - kSequenceNumbers[1]))))); + UnorderedElementsAre(Property( + &ReportBlockData::report_block, + AllOf(Field(&RTCPReportBlock::source_ssrc, kReceiverMainSsrc), + Field(&RTCPReportBlock::sender_ssrc, kSenderSsrc2), + Field(&RTCPReportBlock::fraction_lost, kFracLost[1]), + Field(&RTCPReportBlock::packets_lost, kCumLost[1]), + Field(&RTCPReportBlock::extended_highest_sequence_number, + kSequenceNumbers[1]))))); } TEST(RtcpReceiverTest, GetRtt) { diff --git a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc index 974557ce6e..148e4f973b 100644 --- a/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension_unittest.cc @@ -13,7 +13,6 @@ #include "api/array_view.h" #include "api/transport/rtp/dependency_descriptor.h" #include "common_video/generic_frame_descriptor/generic_frame_info.h" - #include "test/gmock.h" namespace webrtc { diff --git a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc index db077409ee..e05df1a266 100644 --- a/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc +++ b/modules/rtp_rtcp/source/rtp_video_layers_allocation_extension_unittest.cc @@ -13,7 +13,6 @@ #include "api/video/video_layers_allocation.h" #include "rtc_base/bit_buffer.h" #include "rtc_base/buffer.h" - #include "test/gmock.h" namespace webrtc { diff --git a/modules/third_party/g711/g711.h b/modules/third_party/g711/g711.h index 4eef42c0bf..4af819a0af 100644 --- a/modules/third_party/g711/g711.h +++ b/modules/third_party/g711/g711.h @@ -201,7 +201,7 @@ static __inline int bottom_bit(unsigned int bits) { * John Wiley & Sons, pps 98-111 and 472-476. */ -//#define ULAW_ZEROTRAP /* turn on the trap as per the MIL-STD +// #define ULAW_ZEROTRAP /* turn on the trap as per the MIL-STD //*/ #define ULAW_BIAS 0x84 /* Bias for linear code. */ diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc index fb8d55137f..f9be3f83ef 100644 --- a/modules/video_capture/windows/device_info_ds.cc +++ b/modules/video_capture/windows/device_info_ds.cc @@ -259,12 +259,11 @@ IBaseFilter* DeviceInfoDS::GetDeviceFilter(const char* deviceUniqueIdUTF8, deviceFound = true; hr = pM->BindToObject(0, 0, IID_IBaseFilter, (void**)&captureFilter); - if - FAILED(hr) { - RTC_LOG(LS_ERROR) << "Failed to bind to the selected " - "capture device " - << hr; - } + if FAILED (hr) { + RTC_LOG(LS_ERROR) << "Failed to bind to the selected " + "capture device " + << hr; + } if (productUniqueIdUTF8 && productUniqueIdUTF8Length > 0) // Get the device name diff --git a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc index e56e8a92af..7543372e21 100644 --- a/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc +++ b/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc @@ -518,7 +518,6 @@ void VideoCodecTestFixtureImpl::AnalyzeAllFrames( const std::vector* rc_thresholds, const std::vector* quality_thresholds, const BitstreamThresholds* bs_thresholds) { - for (size_t rate_profile_idx = 0; rate_profile_idx < rate_profiles.size(); ++rate_profile_idx) { const size_t first_frame_num = rate_profiles[rate_profile_idx].frame_num; @@ -799,13 +798,12 @@ bool VideoCodecTestFixtureImpl::SetUpAndInitObjects( } } - task_queue->SendTask( - [this]() { - processor_ = std::make_unique( - encoder_.get(), &decoders_, source_frame_reader_.get(), config_, - &stats_, &encoded_frame_writers_, - decoded_frame_writers_.empty() ? nullptr : &decoded_frame_writers_); - }); + task_queue->SendTask([this]() { + processor_ = std::make_unique( + encoder_.get(), &decoders_, source_frame_reader_.get(), config_, + &stats_, &encoded_frame_writers_, + decoded_frame_writers_.empty() ? nullptr : &decoded_frame_writers_); + }); return true; } diff --git a/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/modules/video_coding/codecs/test/videoprocessor_unittest.cc index f1774af5df..40cb5b6395 100644 --- a/modules/video_coding/codecs/test/videoprocessor_unittest.cc +++ b/modules/video_coding/codecs/test/videoprocessor_unittest.cc @@ -51,12 +51,11 @@ class VideoProcessorTest : public ::testing::Test { decoders_.push_back(std::unique_ptr(decoder_mock_)); ExpectInit(); - q_.SendTask( - [this] { - video_processor_ = std::make_unique( - &encoder_mock_, &decoders_, &frame_reader_mock_, config_, &stats_, - &encoded_frame_writers_, /*decoded_frame_writers=*/nullptr); - }); + q_.SendTask([this] { + video_processor_ = std::make_unique( + &encoder_mock_, &decoders_, &frame_reader_mock_, config_, &stats_, + &encoded_frame_writers_, /*decoded_frame_writers=*/nullptr); + }); } ~VideoProcessorTest() { diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc index 9a599cd4bd..52ef5c2d85 100644 --- a/modules/video_coding/packet_buffer.cc +++ b/modules/video_coding/packet_buffer.cc @@ -321,7 +321,7 @@ std::vector> PacketBuffer::FindFrames( // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7106 if (is_h264_descriptor && (buffer_[start_index] == nullptr || - buffer_[start_index]->timestamp != frame_timestamp)) { + buffer_[start_index]->timestamp != frame_timestamp)) { break; } diff --git a/modules/video_coding/rtp_frame_reference_finder.cc b/modules/video_coding/rtp_frame_reference_finder.cc index 3d6543b850..b73fcfe9d5 100644 --- a/modules/video_coding/rtp_frame_reference_finder.cc +++ b/modules/video_coding/rtp_frame_reference_finder.cc @@ -145,8 +145,7 @@ T& RtpFrameReferenceFinderImpl::GetRefFinderAs() { RtpFrameReferenceFinder::RtpFrameReferenceFinder() : RtpFrameReferenceFinder(0) {} -RtpFrameReferenceFinder::RtpFrameReferenceFinder( - int64_t picture_id_offset) +RtpFrameReferenceFinder::RtpFrameReferenceFinder(int64_t picture_id_offset) : picture_id_offset_(picture_id_offset), impl_(std::make_unique()) {} diff --git a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc index 72084a7358..a3cb31ade5 100644 --- a/modules/video_coding/rtp_vp9_ref_finder_unittest.cc +++ b/modules/video_coding/rtp_vp9_ref_finder_unittest.cc @@ -157,8 +157,7 @@ class HasFrameMatcher : public MatcherInterface { public: explicit HasFrameMatcher(int64_t frame_id, const std::vector& expected_refs) - : frame_id_(frame_id), - expected_refs_(expected_refs) {} + : frame_id_(frame_id), expected_refs_(expected_refs) {} bool MatchAndExplain(const FrameVector& frames, MatchResultListener* result_listener) const override { diff --git a/modules/video_coding/timing/jitter_estimator.h b/modules/video_coding/timing/jitter_estimator.h index a89a4bf1fd..89dc64934b 100644 --- a/modules/video_coding/timing/jitter_estimator.h +++ b/modules/video_coding/timing/jitter_estimator.h @@ -172,7 +172,7 @@ class JitterEstimator { // TODO(bugs.webrtc.org/14381): Update `avg_frame_size_bytes_` to DataSize // when api/units have sufficient precision. - double avg_frame_size_bytes_; // Average frame size + double avg_frame_size_bytes_; // Average frame size double var_frame_size_bytes2_; // Frame size variance. Unit is bytes^2. // Largest frame size received (descending with a factor kPsi). // Used by default. diff --git a/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc b/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc index d28052e28d..4e2c759707 100644 --- a/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc +++ b/modules/video_coding/utility/bandwidth_quality_scaler_unittest.cc @@ -97,17 +97,16 @@ class BandwidthQualityScalerTest : scoped_field_trial_(GetParam()), task_queue_("BandwidthQualityScalerTestQueue"), handler_(std::make_unique()) { - task_queue_.SendTask( - [this] { - bandwidth_quality_scaler_ = - std::unique_ptr( - new BandwidthQualityScalerUnderTest(handler_.get())); - bandwidth_quality_scaler_->SetResolutionBitrateLimits( - EncoderInfoSettings:: - GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted()); - // Only for testing. Set first_timestamp_ in RateStatistics to 0. - bandwidth_quality_scaler_->ReportEncodeInfo(0, 0, 0, 0); - }); + task_queue_.SendTask([this] { + bandwidth_quality_scaler_ = + std::unique_ptr( + new BandwidthQualityScalerUnderTest(handler_.get())); + bandwidth_quality_scaler_->SetResolutionBitrateLimits( + EncoderInfoSettings:: + GetDefaultSinglecastBitrateLimitsWhenQpIsUntrusted()); + // Only for testing. Set first_timestamp_ in RateStatistics to 0. + bandwidth_quality_scaler_->ReportEncodeInfo(0, 0, 0, 0); + }); } ~BandwidthQualityScalerTest() { @@ -150,37 +149,35 @@ class BandwidthQualityScalerTest void TriggerBandwidthQualityScalerTest( const std::vector& frame_configs) { - task_queue_.SendTask( - [frame_configs, this] { - RTC_CHECK(!frame_configs.empty()); + task_queue_.SendTask([frame_configs, this] { + RTC_CHECK(!frame_configs.empty()); - int total_frame_nums = 0; - for (const FrameConfig& frame_config : frame_configs) { - total_frame_nums += frame_config.frame_num; - } + int total_frame_nums = 0; + for (const FrameConfig& frame_config : frame_configs) { + total_frame_nums += frame_config.frame_num; + } - EXPECT_EQ( - kFramerateFps * kDefaultBitrateStateUpdateInterval.seconds(), - total_frame_nums); + EXPECT_EQ(kFramerateFps * kDefaultBitrateStateUpdateInterval.seconds(), + total_frame_nums); - uint32_t time_send_to_scaler_ms_ = rtc::TimeMillis(); - for (size_t i = 0; i < frame_configs.size(); ++i) { - const FrameConfig& config = frame_configs[i]; - absl::optional - suitable_bitrate = GetDefaultSuitableBitrateLimit( - config.actual_width * config.actual_height); - EXPECT_TRUE(suitable_bitrate); - for (int j = 0; j <= config.frame_num; ++j) { - time_send_to_scaler_ms_ += kDefaultEncodeTime.ms(); - int frame_size_bytes = - GetFrameSizeBytes(config, suitable_bitrate.value()); - RTC_CHECK(frame_size_bytes > 0); - bandwidth_quality_scaler_->ReportEncodeInfo( - frame_size_bytes, time_send_to_scaler_ms_, - config.actual_width, config.actual_height); - } - } - }); + uint32_t time_send_to_scaler_ms_ = rtc::TimeMillis(); + for (size_t i = 0; i < frame_configs.size(); ++i) { + const FrameConfig& config = frame_configs[i]; + absl::optional suitable_bitrate = + GetDefaultSuitableBitrateLimit(config.actual_width * + config.actual_height); + EXPECT_TRUE(suitable_bitrate); + for (int j = 0; j <= config.frame_num; ++j) { + time_send_to_scaler_ms_ += kDefaultEncodeTime.ms(); + int frame_size_bytes = + GetFrameSizeBytes(config, suitable_bitrate.value()); + RTC_CHECK(frame_size_bytes > 0); + bandwidth_quality_scaler_->ReportEncodeInfo( + frame_size_bytes, time_send_to_scaler_ms_, config.actual_width, + config.actual_height); + } + } + }); } test::ScopedFieldTrials scoped_field_trial_; diff --git a/modules/video_coding/utility/ivf_file_reader_unittest.cc b/modules/video_coding/utility/ivf_file_reader_unittest.cc index c9cf14674b..0e20b7f77c 100644 --- a/modules/video_coding/utility/ivf_file_reader_unittest.cc +++ b/modules/video_coding/utility/ivf_file_reader_unittest.cc @@ -9,11 +9,11 @@ */ #include "modules/video_coding/utility/ivf_file_reader.h" -#include "modules/video_coding/utility/ivf_file_writer.h" #include #include +#include "modules/video_coding/utility/ivf_file_writer.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" diff --git a/modules/video_coding/utility/quality_scaler_unittest.cc b/modules/video_coding/utility/quality_scaler_unittest.cc index c17159fb64..50410dd25b 100644 --- a/modules/video_coding/utility/quality_scaler_unittest.cc +++ b/modules/video_coding/utility/quality_scaler_unittest.cc @@ -72,11 +72,10 @@ class QualityScalerTest : public ::testing::Test, : scoped_field_trial_(GetParam()), task_queue_("QualityScalerTestQueue"), handler_(std::make_unique()) { - task_queue_.SendTask( - [this] { - qs_ = std::unique_ptr(new QualityScalerUnderTest( - handler_.get(), VideoEncoder::QpThresholds(kLowQp, kHighQp))); - }); + task_queue_.SendTask([this] { + qs_ = std::unique_ptr(new QualityScalerUnderTest( + handler_.get(), VideoEncoder::QpThresholds(kLowQp, kHighQp))); + }); } ~QualityScalerTest() override {