Attachment 347795 Details for Bug 188745 – WIP GTK compilation patch

[patch] WIP GTK compilation patch

WIP.patch (text/plain), 44.75 KB, created by Alejandro G. Castro on 2018-08-22 08:20:58 PDT

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Filename:

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Creator: Alejandro G. Castro

Created: 2018-08-22 08:20:58 PDT

Size: 44.75 KB

patch

obsolete

>diff --git a/Source/ThirdParty/libwebrtc/CMakeLists.txt b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>index 550cf2a16de..5552a6ef63f 100644
>--- a/Source/ThirdParty/libwebrtc/CMakeLists.txt
>+++ b/Source/ThirdParty/libwebrtc/CMakeLists.txt
>@@ -17,6 +17,11 @@ if (NOT ALSALIB_FOUND)
> endif ()
> 
> set(webrtc_SOURCES
>+    Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc
>+    Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc
>+    Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc
>+    Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc
>+    Source/third_party/abseil-cpp/absl/types/optional.cc
>     Source/third_party/boringssl/err_data.c
>     Source/third_party/boringssl/src/crypto/asn1/a_bitstr.c
>     Source/third_party/boringssl/src/crypto/asn1/a_bool.c
>@@ -294,17 +299,14 @@ set(webrtc_SOURCES
>     Source/third_party/libyuv/source/rotate_any.cc
>     Source/third_party/libyuv/source/rotate_argb.cc
>     Source/third_party/libyuv/source/rotate_common.cc
>-    Source/third_party/libyuv/source/rotate_dspr2.cc
>     Source/third_party/libyuv/source/rotate_gcc.cc
>     Source/third_party/libyuv/source/row_any.cc
>     Source/third_party/libyuv/source/row_common.cc
>-    Source/third_party/libyuv/source/row_dspr2.cc
>     Source/third_party/libyuv/source/row_gcc.cc
>     Source/third_party/libyuv/source/scale.cc
>     Source/third_party/libyuv/source/scale_any.cc
>     Source/third_party/libyuv/source/scale_argb.cc
>     Source/third_party/libyuv/source/scale_common.cc
>-    Source/third_party/libyuv/source/scale_dspr2.cc
>     Source/third_party/libyuv/source/scale_gcc.cc
>     Source/third_party/libyuv/source/video_common.cc
>     Source/third_party/opus/src/celt/bands.c
>@@ -442,6 +444,8 @@ set(webrtc_SOURCES
>     Source/third_party/opus/src/src/opus_multistream_decoder.c
>     Source/third_party/opus/src/src/opus_multistream_encoder.c
>     Source/third_party/opus/src/src/repacketizer.c
>+    Source/third_party/rnnoise/src/kiss_fft.cc
>+    Source/third_party/rnnoise/src/rnn_vad_weights.cc
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_asconf.c
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_auth.c
>     Source/third_party/usrsctp/usrsctplib/netinet/sctp_bsd_addr.c
>@@ -465,8 +469,10 @@ set(webrtc_SOURCES
>     Source/third_party/usrsctp/usrsctplib/user_mbuf.c
>     Source/third_party/usrsctp/usrsctplib/user_recv_thread.c
>     Source/third_party/usrsctp/usrsctplib/user_socket.c
>+    Source/webrtc/api/audio/audio_frame.cc
>     Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>     Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>+    Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>     Source/webrtc/api/audio_codecs/audio_decoder.cc
>     Source/webrtc/api/audio_codecs/audio_encoder.cc
>     Source/webrtc/api/audio_codecs/audio_format.cc
>@@ -485,46 +491,87 @@ set(webrtc_SOURCES
>     Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>     Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc
>     Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
>+    Source/webrtc/api/audio_options.cc
>+    Source/webrtc/api/call/transport.cc
>     Source/webrtc/api/candidate.cc
>+    Source/webrtc/api/datachannelinterface.cc
>     Source/webrtc/api/jsep.cc
>+    Source/webrtc/api/jsepicecandidate.cc
>     Source/webrtc/api/mediaconstraintsinterface.cc
>     Source/webrtc/api/mediastreaminterface.cc
>     Source/webrtc/api/mediatypes.cc
>-    Source/webrtc/api/optional.cc
>+    Source/webrtc/api/peerconnectioninterface.cc
>     Source/webrtc/api/proxy.cc
>     Source/webrtc/api/rtcerror.cc
>     Source/webrtc/api/rtp_headers.cc
>     Source/webrtc/api/rtpparameters.cc
>+    Source/webrtc/api/rtpreceiverinterface.cc
>+    Source/webrtc/api/rtptransceiverinterface.cc
>     Source/webrtc/api/statstypes.cc
>-    Source/webrtc/api/umametrics.cc
>+    Source/webrtc/api/transport/bitrate_settings.cc
>+    Source/webrtc/api/transport/network_types.cc
>+    Source/webrtc/api/units/data_rate.cc
>+    Source/webrtc/api/units/data_size.cc
>+    Source/webrtc/api/units/time_delta.cc
>+    Source/webrtc/api/units/timestamp.cc
>+    Source/webrtc/api/video/color_space.cc
>+    Source/webrtc/api/video/encoded_frame.cc
>+    Source/webrtc/api/video/i010_buffer.cc
>     Source/webrtc/api/video/i420_buffer.cc
>+    Source/webrtc/api/video/video_bitrate_allocation.cc
>     Source/webrtc/api/video/video_content_type.cc
>     Source/webrtc/api/video/video_frame.cc
>     Source/webrtc/api/video/video_frame_buffer.cc
>+    Source/webrtc/api/video/video_source_interface.cc
>+    Source/webrtc/api/video/video_stream_decoder_create.cc
>+    Source/webrtc/api/video/video_stream_encoder_create.cc
>     Source/webrtc/api/video/video_timing.cc
>+    Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc
>+    Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>+    Source/webrtc/api/video_codecs/sdp_video_format.cc
>+    Source/webrtc/api/video_codecs/video_codec.cc
>+    Source/webrtc/api/video_codecs/video_decoder.cc
>+    Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc
>     Source/webrtc/api/video_codecs/video_encoder.cc
>-    Source/webrtc/api/videosourceinterface.cc
>+    Source/webrtc/api/video_codecs/video_encoder_config.cc
>+    Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>+    Source/webrtc/audio/audio_level.cc
>     Source/webrtc/audio/audio_receive_stream.cc
>     Source/webrtc/audio/audio_send_stream.cc
>     Source/webrtc/audio/audio_state.cc
>     Source/webrtc/audio/audio_transport_impl.cc
>+    Source/webrtc/audio/channel.cc
>+    Source/webrtc/audio/channel_proxy.cc
>     Source/webrtc/audio/null_audio_poller.cc
>+    Source/webrtc/audio/remix_resample.cc
>     Source/webrtc/audio/time_interval.cc
>+    Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>     Source/webrtc/audio/utility/audio_frame_operations.cc
>+    Source/webrtc/call/audio_receive_stream.cc
>     Source/webrtc/call/audio_send_stream.cc
>+    Source/webrtc/call/audio_state.cc
>     Source/webrtc/call/bitrate_allocator.cc
>     Source/webrtc/call/call.cc
>+    Source/webrtc/call/call_config.cc
>     Source/webrtc/call/callfactory.cc
>+    Source/webrtc/call/degraded_call.cc
>+    Source/webrtc/call/fake_network_pipe.cc
>+    Source/webrtc/call/flexfec_receive_stream.cc
>     Source/webrtc/call/flexfec_receive_stream_impl.cc
>+    Source/webrtc/call/packet_receiver.cc
>+    Source/webrtc/call/receive_time_calculator.cc
>     Source/webrtc/call/rtcp_demuxer.cc
>+    Source/webrtc/call/rtp_bitrate_configurator.cc
>     Source/webrtc/call/rtp_config.cc
>     Source/webrtc/call/rtp_demuxer.cc
>+    Source/webrtc/call/rtp_payload_params.cc
>     Source/webrtc/call/rtp_rtcp_demuxer_helper.cc
>     Source/webrtc/call/rtp_stream_receiver_controller.cc
>     Source/webrtc/call/rtp_transport_controller_send.cc
>+    Source/webrtc/call/rtp_video_sender.cc
>     Source/webrtc/call/rtx_receive_stream.cc
>+    Source/webrtc/call/simulated_network.cc
>     Source/webrtc/call/syncable.cc
>-    Source/webrtc/call/video_config.cc
>     Source/webrtc/call/video_receive_stream.cc
>     Source/webrtc/call/video_send_stream.cc
>     Source/webrtc/common_audio/audio_converter.cc
>@@ -532,7 +579,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/audio_util.cc
>     Source/webrtc/common_audio/blocker.cc
>     Source/webrtc/common_audio/channel_buffer.cc
>-    Source/webrtc/common_audio/fft4g.c
>     Source/webrtc/common_audio/fir_filter_c.cc
>     Source/webrtc/common_audio/fir_filter_factory.cc
>     Source/webrtc/common_audio/lapped_transform.cc
>@@ -573,12 +619,13 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/signal_processing/spl_init.c
>     Source/webrtc/common_audio/signal_processing/spl_inl.c
>     Source/webrtc/common_audio/signal_processing/spl_sqrt.c
>-    Source/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
>     Source/webrtc/common_audio/signal_processing/splitting_filter.c
>     Source/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
>     Source/webrtc/common_audio/signal_processing/vector_scaling_operations.c
>     Source/webrtc/common_audio/smoothing_filter.cc
>     Source/webrtc/common_audio/sparse_fir_filter.cc
>+    Source/webrtc/common_audio/third_party/fft4g/fft4g.c
>+    Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
>     Source/webrtc/common_audio/vad/vad.cc
>     Source/webrtc/common_audio/vad/vad_core.c
>     Source/webrtc/common_audio/vad/vad_filterbank.c
>@@ -588,7 +635,6 @@ set(webrtc_SOURCES
>     Source/webrtc/common_audio/wav_file.cc
>     Source/webrtc/common_audio/wav_header.cc
>     Source/webrtc/common_audio/window_generator.cc
>-    Source/webrtc/common_types.cc
>     Source/webrtc/common_video/bitrate_adjuster.cc
>     Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>     Source/webrtc/common_video/h264/h264_common.cc
>@@ -608,6 +654,8 @@ set(webrtc_SOURCES
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>+    Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>+    Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>@@ -617,13 +665,16 @@ set(webrtc_SOURCES
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>     Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>+    Source/webrtc/logging/rtc_event_log/icelogger.cc
>     Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file.cc
>     Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>     Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc
>+    Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>     Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc
>     Source/webrtc/media/base/adaptedvideotracksource.cc
>     Source/webrtc/media/base/codec.cc
>     Source/webrtc/media/base/h264_profile_level_id.cc
>+    Source/webrtc/media/base/mediachannel.cc
>     Source/webrtc/media/base/mediaconstants.cc
>     Source/webrtc/media/base/mediaengine.cc
>     Source/webrtc/media/base/rtpdataengine.cc
>@@ -646,8 +697,6 @@ set(webrtc_SOURCES
>     Source/webrtc/media/engine/scopedvideoencoder.cc
>     Source/webrtc/media/engine/simulcast.cc
>     Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>-    Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
>-    Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.cc
>     Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>     Source/webrtc/media/engine/webrtcmediaengine.cc
>     Source/webrtc/media/engine/webrtcvideocapturer.cc
>@@ -681,12 +730,9 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>     Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>     Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>-    Source/webrtc/modules/audio_coding/codecs/g711/g711.c
>     Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
>     Source/webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
>     Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>-    Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
>-    Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
>     Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
>     Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
>     Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
>@@ -802,12 +848,11 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>-    Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
>-    Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
>+    Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
>     Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
>@@ -838,8 +883,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>     Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc
>     Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>-    Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
>-    Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
>     Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>     Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>     Source/webrtc/modules/audio_coding/neteq/delay_peak_detector.cc
>@@ -847,6 +890,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
>     Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>     Source/webrtc/modules/audio_coding/neteq/expand.cc
>+    Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.cc
>     Source/webrtc/modules/audio_coding/neteq/merge.cc
>     Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>     Source/webrtc/modules/audio_coding/neteq/neteq.cc
>@@ -911,7 +955,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>     Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>     Source/webrtc/modules/audio_processing/aec3/matrix_buffer.cc
>-    Source/webrtc/modules/audio_processing/aec3/output_selector.cc
>     Source/webrtc/modules/audio_processing/aec3/render_buffer.cc
>     Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>     Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>@@ -929,38 +972,54 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>     Source/webrtc/modules/audio_processing/agc/agc.cc
>     Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>+    Source/webrtc/modules/audio_processing/agc/utility.cc
>     Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.c
>     Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
>     Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>-    Source/webrtc/modules/audio_processing/agc/utility.cc
>-    Source/webrtc/modules/audio_processing/agc2/gain_controller2.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>+    Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>+    Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>+    Source/webrtc/modules/audio_processing/agc2/down_sampler.cc
>+    Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>+    Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>+    Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>+    Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>+    Source/webrtc/modules/audio_processing/agc2/limiter.cc
>+    Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
>+    Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>+    Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>+    Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>+    Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>+    Source/webrtc/modules/audio_processing/agc2/vector_float_frame.cc
>     Source/webrtc/modules/audio_processing/audio_buffer.cc
>     Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>-    Source/webrtc/modules/audio_processing/beamformer/array_util.cc
>-    Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
>-    Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
>     Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>+    Source/webrtc/modules/audio_processing/echo_cancellation_proxy.cc
>     Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>+    Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.cc
>     Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc
>     Source/webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc
>     Source/webrtc/modules/audio_processing/echo_detector/moving_max.cc
>     Source/webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc
>     Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>     Source/webrtc/modules/audio_processing/gain_control_impl.cc
>+    Source/webrtc/modules/audio_processing/gain_controller2.cc
>     Source/webrtc/modules/audio_processing/include/aec_dump.cc
>     Source/webrtc/modules/audio_processing/include/audio_processing.cc
>     Source/webrtc/modules/audio_processing/include/audio_processing_statistics.cc
>     Source/webrtc/modules/audio_processing/include/config.cc
>-    Source/webrtc/modules/audio_processing/level_controller/biquad_filter.cc
>-    Source/webrtc/modules/audio_processing/level_controller/down_sampler.cc
>-    Source/webrtc/modules/audio_processing/level_controller/gain_applier.cc
>-    Source/webrtc/modules/audio_processing/level_controller/gain_selector.cc
>-    Source/webrtc/modules/audio_processing/level_controller/level_controller.cc
>-    Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.cc
>-    Source/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
>     Source/webrtc/modules/audio_processing/level_estimator_impl.cc
>     Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>     Source/webrtc/modules/audio_processing/low_cut_filter.cc
>@@ -992,25 +1051,37 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>     Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>     Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>-    Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/delay_based_bwe.cc
>-    Source/webrtc/modules/congestion_controller/median_slope_estimator.cc
>-    Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.cc
>-    Source/webrtc/modules/congestion_controller/probe_controller.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>+    Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>+    Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.cc
>+    Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.cc
>+    Source/webrtc/modules/congestion_controller/bbr/rtt_stats.cc
>+    Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>+    Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
>     Source/webrtc/modules/congestion_controller/receive_side_congestion_controller.cc
>+    Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>+    Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>+    Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>+    Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>     Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>     Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>-    Source/webrtc/modules/congestion_controller/trendline_estimator.cc
>-    Source/webrtc/modules/media_file/media_file_impl.cc
>-    Source/webrtc/modules/media_file/media_file_utility.cc
>-    Source/webrtc/modules/pacing/alr_detector.cc
>     Source/webrtc/modules/pacing/bitrate_prober.cc
>     Source/webrtc/modules/pacing/interval_budget.cc
>     Source/webrtc/modules/pacing/paced_sender.cc
>     Source/webrtc/modules/pacing/packet_queue.cc
>-    Source/webrtc/modules/pacing/packet_queue2.cc
>+    Source/webrtc/modules/pacing/packet_queue_interface.cc
>     Source/webrtc/modules/pacing/packet_router.cc
>+    Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>     Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>     Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>     Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>@@ -1020,9 +1091,11 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>     Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>     Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
>-    Source/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
>     Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>+    Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>     Source/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc
>+    Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
>+    Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver.cc
>     Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>@@ -1067,15 +1140,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>-    Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
>@@ -1086,11 +1161,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>     Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>+    Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>     Source/webrtc/modules/rtp_rtcp/source/time_util.cc
>     Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.cc
>     Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>+    Source/webrtc/modules/third_party/fft/fft.c
>+    Source/webrtc/modules/third_party/g711/g711.c
>+    Source/webrtc/modules/third_party/g722/g722_decode.c
>+    Source/webrtc/modules/third_party/g722/g722_encode.c
>+    Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.c
>     Source/webrtc/modules/utility/source/process_thread_impl.cc
>     Source/webrtc/modules/video_capture/device_info_impl.cc
>     Source/webrtc/modules/video_capture/external/device_info_external.cc
>@@ -1099,20 +1180,17 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_capture/linux/video_capture_linux.cc
>     Source/webrtc/modules/video_capture/video_capture_factory.cc
>     Source/webrtc/modules/video_capture/video_capture_impl.cc
>-    Source/webrtc/modules/video_coding/codec_database.cc
>     Source/webrtc/modules/video_coding/codec_timer.cc
>     Source/webrtc/modules/video_coding/codecs/h264/h264.cc
>     Source/webrtc/modules/video_coding/codecs/i420/i420.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>     Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.cc
>-    Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>+    Source/webrtc/modules/video_coding/decoder_database.cc
>     Source/webrtc/modules/video_coding/decoding_state.cc
>     Source/webrtc/modules/video_coding/encoded_frame.cc
>+    Source/webrtc/modules/video_coding/encoder_database.cc
>+    Source/webrtc/modules/video_coding/fec_controller_default.cc
>     Source/webrtc/modules/video_coding/frame_buffer.cc
>     Source/webrtc/modules/video_coding/frame_buffer2.cc
>     Source/webrtc/modules/video_coding/frame_object.cc
>@@ -1129,7 +1207,6 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_coding/nack_module.cc
>     Source/webrtc/modules/video_coding/packet.cc
>     Source/webrtc/modules/video_coding/packet_buffer.cc
>-    Source/webrtc/modules/video_coding/protection_bitrate_calculator.cc
>     Source/webrtc/modules/video_coding/qp_parser.cc
>     Source/webrtc/modules/video_coding/receiver.cc
>     Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
>@@ -1142,9 +1219,12 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc
>     Source/webrtc/modules/video_coding/utility/moving_average.cc
>     Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>+    Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
>+    Source/webrtc/modules/video_coding/utility/simulcast_utility.cc
>     Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc
>     Source/webrtc/modules/video_coding/utility/vp9_uncompressed_header_parser.cc
>     Source/webrtc/modules/video_coding/video_codec_initializer.cc
>+    Source/webrtc/modules/video_coding/video_coding_defines.cc
>     Source/webrtc/modules/video_coding/video_coding_impl.cc
>     Source/webrtc/modules/video_coding/video_receiver.cc
>     Source/webrtc/modules/video_coding/video_sender.cc
>@@ -1154,6 +1234,7 @@ set(webrtc_SOURCES
>     Source/webrtc/modules/video_processing/util/skin_detection.cc
>     Source/webrtc/modules/video_processing/video_denoiser.cc
>     Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>+    Source/webrtc/p2p/base/basicasyncresolverfactory.cc
>     Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>     Source/webrtc/p2p/base/dtlstransport.cc
>     Source/webrtc/p2p/base/dtlstransportinternal.cc
>@@ -1167,9 +1248,9 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/portallocator.cc
>     Source/webrtc/p2p/base/portinterface.cc
>     Source/webrtc/p2p/base/pseudotcp.cc
>+    Source/webrtc/p2p/base/regatheringcontroller.cc
>     Source/webrtc/p2p/base/relayport.cc
>     Source/webrtc/p2p/base/relayserver.cc
>-    Source/webrtc/p2p/base/session.cc
>     Source/webrtc/p2p/base/stun.cc
>     Source/webrtc/p2p/base/stunport.cc
>     Source/webrtc/p2p/base/stunrequest.cc
>@@ -1181,16 +1262,13 @@ set(webrtc_SOURCES
>     Source/webrtc/p2p/base/turnserver.cc
>     Source/webrtc/p2p/base/udptransport.cc
>     Source/webrtc/p2p/client/basicportallocator.cc
>-    Source/webrtc/p2p/client/socketmonitor.cc
>     Source/webrtc/p2p/client/turnportfactory.cc
>     Source/webrtc/p2p/stunprober/stunprober.cc
>-    Source/webrtc/pc/audiomonitor.cc
>     Source/webrtc/pc/audiotrack.cc
>     Source/webrtc/pc/bundlefilter.cc
>     Source/webrtc/pc/channel.cc
>     Source/webrtc/pc/channelmanager.cc
>     Source/webrtc/pc/createpeerconnectionfactory.cc
>-    Source/webrtc/pc/currentspeakermonitor.cc
>     Source/webrtc/pc/datachannel.cc
>     Source/webrtc/pc/dtlssrtptransport.cc
>     Source/webrtc/pc/dtmfsender.cc
>@@ -1199,8 +1277,8 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/jsepicecandidate.cc
>     Source/webrtc/pc/jsepsessiondescription.cc
>     Source/webrtc/pc/jseptransport.cc
>+    Source/webrtc/pc/jseptransportcontroller.cc
>     Source/webrtc/pc/localaudiosource.cc
>-    Source/webrtc/pc/mediamonitor.cc
>     Source/webrtc/pc/mediasession.cc
>     Source/webrtc/pc/mediastream.cc
>     Source/webrtc/pc/mediastreamobserver.cc
>@@ -1209,7 +1287,9 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/remoteaudiosource.cc
>     Source/webrtc/pc/rtcpmuxfilter.cc
>     Source/webrtc/pc/rtcstatscollector.cc
>+    Source/webrtc/pc/rtcstatstraversal.cc
>     Source/webrtc/pc/rtpmediautils.cc
>+    Source/webrtc/pc/rtpparametersconversion.cc
>     Source/webrtc/pc/rtpreceiver.cc
>     Source/webrtc/pc/rtpsender.cc
>     Source/webrtc/pc/rtptransceiver.cc
>@@ -1222,7 +1302,7 @@ set(webrtc_SOURCES
>     Source/webrtc/pc/srtptransport.cc
>     Source/webrtc/pc/statscollector.cc
>     Source/webrtc/pc/trackmediainfomap.cc
>-    Source/webrtc/pc/transportcontroller.cc
>+    Source/webrtc/pc/transportstats.cc
>     Source/webrtc/pc/videocapturertracksource.cc
>     Source/webrtc/pc/videotrack.cc
>     Source/webrtc/pc/videotracksource.cc
>@@ -1234,7 +1314,6 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/asyncsocket.cc
>     Source/webrtc/rtc_base/asynctcpsocket.cc
>     Source/webrtc/rtc_base/asyncudpsocket.cc
>-    Source/webrtc/rtc_base/base64.cc
>     Source/webrtc/rtc_base/bitbuffer.cc
>     Source/webrtc/rtc_base/bitrateallocationstrategy.cc
>     Source/webrtc/rtc_base/bufferqueue.cc
>@@ -1245,9 +1324,14 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/crc32.cc
>     Source/webrtc/rtc_base/criticalsection.cc
>     Source/webrtc/rtc_base/cryptstring.cc
>+    Source/webrtc/rtc_base/data_rate_limiter.cc
>     Source/webrtc/rtc_base/event.cc
>     Source/webrtc/rtc_base/event_tracer.cc
>     Source/webrtc/rtc_base/experiments/alr_experiment.cc
>+    Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc
>+    Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>+    Source/webrtc/rtc_base/experiments/field_trial_units.cc
>+    Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>     Source/webrtc/rtc_base/file.cc
>     Source/webrtc/rtc_base/file_posix.cc
>     Source/webrtc/rtc_base/filerotatingstream.cc
>@@ -1264,8 +1348,7 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/location.cc
>     Source/webrtc/rtc_base/logging.cc
>     Source/webrtc/rtc_base/logsinks.cc
>-    Source/webrtc/rtc_base/md5.cc
>-    Source/webrtc/rtc_base/md5digest.cc
>+    Source/webrtc/rtc_base/memory/aligned_malloc.cc
>     Source/webrtc/rtc_base/memory_usage.cc
>     Source/webrtc/rtc_base/messagedigest.cc
>     Source/webrtc/rtc_base/messagehandler.cc
>@@ -1281,52 +1364,63 @@ set(webrtc_SOURCES
>     Source/webrtc/rtc_base/nullsocketserver.cc
>     Source/webrtc/rtc_base/numerics/exp_filter.cc
>     Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>+    Source/webrtc/rtc_base/numerics/sample_counter.cc
>     Source/webrtc/rtc_base/openssladapter.cc
>+    Source/webrtc/rtc_base/opensslcertificate.cc
>     Source/webrtc/rtc_base/openssldigest.cc
>     Source/webrtc/rtc_base/opensslidentity.cc
>+    Source/webrtc/rtc_base/opensslsessioncache.cc
>     Source/webrtc/rtc_base/opensslstreamadapter.cc
>+    Source/webrtc/rtc_base/opensslutility.cc
>     Source/webrtc/rtc_base/optionsfile.cc
>     Source/webrtc/rtc_base/pathutils.cc
>     Source/webrtc/rtc_base/physicalsocketserver.cc
>     Source/webrtc/rtc_base/platform_file.cc
>     Source/webrtc/rtc_base/platform_thread.cc
>+    Source/webrtc/rtc_base/platform_thread_types.cc
>     Source/webrtc/rtc_base/proxyinfo.cc
>     Source/webrtc/rtc_base/proxyserver.cc
>     Source/webrtc/rtc_base/race_checker.cc
>     Source/webrtc/rtc_base/random.cc
>     Source/webrtc/rtc_base/rate_limiter.cc
>     Source/webrtc/rtc_base/rate_statistics.cc
>-    Source/webrtc/rtc_base/ratelimiter.cc
>     Source/webrtc/rtc_base/ratetracker.cc
>     Source/webrtc/rtc_base/rtccertificate.cc
>     Source/webrtc/rtc_base/rtccertificategenerator.cc
>     Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>-    Source/webrtc/rtc_base/sha1.cc
>-    Source/webrtc/rtc_base/sha1digest.cc
>     Source/webrtc/rtc_base/signalthread.cc
>-    Source/webrtc/rtc_base/sigslot.cc
>+    Source/webrtc/rtc_base/socket.cc
>     Source/webrtc/rtc_base/socketadapters.cc
>     Source/webrtc/rtc_base/socketaddress.cc
>     Source/webrtc/rtc_base/socketaddresspair.cc
>     Source/webrtc/rtc_base/socketstream.cc
>     Source/webrtc/rtc_base/ssladapter.cc
>+    Source/webrtc/rtc_base/sslcertificate.cc
>     Source/webrtc/rtc_base/sslfingerprint.cc
>     Source/webrtc/rtc_base/sslidentity.cc
>     Source/webrtc/rtc_base/sslstreamadapter.cc
>     Source/webrtc/rtc_base/stream.cc
>     Source/webrtc/rtc_base/string_to_number.cc
>     Source/webrtc/rtc_base/stringencode.cc
>+    Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>+    Source/webrtc/rtc_base/strings/string_builder.cc
>     Source/webrtc/rtc_base/stringutils.cc
>+    Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>+    Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.cc
>+    Source/webrtc/rtc_base/system/file_wrapper.cc
>     Source/webrtc/rtc_base/task_queue_libevent.cc
>     Source/webrtc/rtc_base/task_queue_posix.cc
>+    Source/webrtc/rtc_base/third_party/base64/base64.cc
>+    Source/webrtc/rtc_base/third_party/sigslot/sigslot.cc
>     Source/webrtc/rtc_base/thread.cc
>     Source/webrtc/rtc_base/thread_checker_impl.cc
>+    Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>     Source/webrtc/rtc_base/timestampaligner.cc
>     Source/webrtc/rtc_base/timeutils.cc
>-    Source/webrtc/rtc_base/transformadapter.cc
>     Source/webrtc/rtc_base/unixfilesystem.cc
>     Source/webrtc/rtc_base/virtualsocketserver.cc
>     Source/webrtc/rtc_base/weak_ptr.cc
>+    Source/webrtc/rtc_base/zero_memory.cc
>     Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc
>     Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>     Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc
>@@ -1334,7 +1428,6 @@ set(webrtc_SOURCES
>     Source/webrtc/stats/rtcstats.cc
>     Source/webrtc/stats/rtcstats_objects.cc
>     Source/webrtc/stats/rtcstatsreport.cc
>-    Source/webrtc/system_wrappers/source/aligned_malloc.cc
>     Source/webrtc/system_wrappers/source/clock.cc
>     Source/webrtc/system_wrappers/source/cpu_features.cc
>     Source/webrtc/system_wrappers/source/cpu_features_linux.c
>@@ -1342,18 +1435,13 @@ set(webrtc_SOURCES
>     Source/webrtc/system_wrappers/source/event.cc
>     Source/webrtc/system_wrappers/source/event_timer_posix.cc
>     Source/webrtc/system_wrappers/source/field_trial_default.cc
>-    Source/webrtc/system_wrappers/source/file_impl.cc
>     Source/webrtc/system_wrappers/source/metrics_default.cc
>     Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>     Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc
>-    Source/webrtc/system_wrappers/source/rw_lock.cc
>-    Source/webrtc/system_wrappers/source/rw_lock_posix.cc
>     Source/webrtc/system_wrappers/source/sleep.cc
>-    Source/webrtc/system_wrappers/source/timestamp_extrapolator.cc
>     Source/webrtc/video/call_stats.cc
>     Source/webrtc/video/encoder_rtcp_feedback.cc
>     Source/webrtc/video/overuse_frame_detector.cc
>-    Source/webrtc/video/payload_router.cc
>     Source/webrtc/video/quality_threshold.cc
>     Source/webrtc/video/receive_statistics_proxy.cc
>     Source/webrtc/video/report_block_stats.cc
>@@ -1364,15 +1452,13 @@ set(webrtc_SOURCES
>     Source/webrtc/video/stats_counter.cc
>     Source/webrtc/video/stream_synchronization.cc
>     Source/webrtc/video/transport_adapter.cc
>+    Source/webrtc/video/video_quality_observer.cc
>     Source/webrtc/video/video_receive_stream.cc
>     Source/webrtc/video/video_send_stream.cc
>+    Source/webrtc/video/video_send_stream_impl.cc
>     Source/webrtc/video/video_stream_decoder.cc
>+    Source/webrtc/video/video_stream_decoder_impl.cc
>     Source/webrtc/video/video_stream_encoder.cc
>-    Source/webrtc/voice_engine/audio_level.cc
>-    Source/webrtc/voice_engine/channel.cc
>-    Source/webrtc/voice_engine/channel_proxy.cc
>-    Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
>-    Source/webrtc/voice_engine/utility.cc
>     $<TARGET_OBJECTS:libsrtp>
> )
> 
>@@ -1429,6 +1515,8 @@ target_compile_definitions(webrtc PRIVATE
>   WEBRTC_USE_BUILTIN_ISAC_FIX=1
>   WEBRTC_USE_BUILTIN_ISAC_FLOAT=0
>   WTF_USE_DYNAMIC_ANNOTATIONS=1
>+  LINUX_ALSA
>+  RTC_DISABLE_VP9
>   _GNU_SOURCE
>   __Userspace__
>   __Userspace_os_Linux
>@@ -1436,6 +1524,7 @@ target_compile_definitions(webrtc PRIVATE
> 
> target_include_directories(webrtc PRIVATE
>     Source
>+    Source/third_party/abseil-cpp
>     Source/third_party/boringssl/src/include
>     Source/third_party/jsoncpp/overrides/include
>     Source/third_party/jsoncpp/source/include
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>index d3b1a7c5cb1..cbed3334d77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>@@ -16,7 +16,14 @@
> #include <unistd.h>
> #include <list>
> 
>+#if defined(WEBRTC_LINUX)
>+#include <event2/event.h>
>+#include <event2/event_compat.h>
>+#include <event2/event_struct.h>
>+#else
> #include "base/third_party/libevent/event.h"
>+#endif
>+
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/WTF/wtf/DisallowCType.h b/Source/WTF/wtf/DisallowCType.h
>index 30544e729fd..386eb328e03 100644
>--- a/Source/WTF/wtf/DisallowCType.h
>+++ b/Source/WTF/wtf/DisallowCType.h
>@@ -40,7 +40,7 @@
> // are used from wx headers. On GTK+ for Mac many GTK+ files include <libintl.h>
> // or <glib/gi18n-lib.h>, which in turn include <xlocale/_ctype.h> which uses
> // isacii(). 
>-#if !(OS(DARWIN) && PLATFORM(GTK)) && !defined(_LIBCPP_VERSION) && defined(__GLIBC__)
>+#if !(OS(DARWIN) && PLATFORM(GTK)) && !PLATFORM(WPE) && !PLATFORM(GTK) && !defined(_LIBCPP_VERSION) && defined(__GLIBC__)
> 
> #include <ctype.h>
> 
>diff --git a/Source/WebCore/CMakeLists.txt b/Source/WebCore/CMakeLists.txt
>index 2f04d538df0..a9c4a5f4347 100644
>--- a/Source/WebCore/CMakeLists.txt
>+++ b/Source/WebCore/CMakeLists.txt
>@@ -1554,7 +1554,8 @@ endif ()
> 
> if (USE_LIBWEBRTC)
>   list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES "${THIRDPARTY_DIR}/libwebrtc/Source/"
>-      "${THIRDPARTY_DIR}/libwebrtc/Source/webrtc")
>+      "${THIRDPARTY_DIR}/libwebrtc/Source/webrtc"
>+      "${THIRDPARTY_DIR}/libwebrtc/Source/third_party/abseil-cpp")
>   list(APPEND WebCore_LIBRARIES webrtc)
>   list(APPEND WebCore_SOURCES
>       Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.cpp
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>index 0ec2f754e7c..096ded08022 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoDecoderFactory.cpp
>@@ -143,7 +143,6 @@ public:
> 
>     int32_t Decode(const webrtc::EncodedImage& inputImage,
>         bool,
>-        const webrtc::RTPFragmentationHeader*,
>         const webrtc::CodecSpecificInfo*,
>         int64_t renderTimeMs) override
>     {
>@@ -233,7 +232,7 @@ public:
>         GST_LOG_OBJECT(pipeline(), "Output decoded frame! %d -> %" GST_PTR_FORMAT,
>             frame->timestamp(), buffer);
> 
>-        m_imageReadyCb->Decoded(*frame.get(), rtc::Optional<int32_t>(), rtc::Optional<uint8_t>());
>+        m_imageReadyCb->Decoded(*frame.get(), absl::optional<int32_t>(), absl::optional<uint8_t>());
> 
>         return GST_FLOW_OK;
>     }
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>index 21649c25664..8037b93b79b 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/GStreamerVideoEncoderFactory.cpp
>@@ -61,15 +61,13 @@ static GRefPtr<GRegex> bitrateKBitPerSec;
> class GStreamerVideoEncoder : public webrtc::VideoEncoder {
> public:
>     GStreamerVideoEncoder(const webrtc::SdpVideoFormat&)
>-        : m_pictureId(0)
>-        , m_firstFramePts(GST_CLOCK_TIME_NONE)
>+        : m_firstFramePts(GST_CLOCK_TIME_NONE)
>         , m_restrictionCaps(adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>         , m_bitrateSetter(nullptr)
>     {
>     }
>     GStreamerVideoEncoder()
>-        : m_pictureId(0)
>-        , m_firstFramePts(GST_CLOCK_TIME_NONE)
>+        : m_firstFramePts(GST_CLOCK_TIME_NONE)
>         , m_restrictionCaps(adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>         , m_bitrateSetter(nullptr)
>     {
>@@ -241,7 +239,6 @@ public:
>         PopulateCodecSpecific(&codecSpecifiInfos, buffer);
> 
>         webrtc::EncodedImageCallback::Result result = m_imageReadyCb->OnEncodedImage(frame, &codecSpecifiInfos, fragmentationInfo);
>-        m_pictureId = (m_pictureId + 1) & 0x7FFF;
>         if (result.error != webrtc::EncodedImageCallback::Result::OK) {
>             GST_ELEMENT_ERROR(m_pipeline.get(), LIBRARY, FAILED, (nullptr),
>                 ("Encode callback failed: %d", result.error));
>@@ -391,9 +388,6 @@ public:
>         m_restrictionCaps = caps;
>     }
> 
>-protected:
>-    int16_t m_pictureId;
>-
> private:
>     static GstFlowReturn newSampleCallbackTramp(GstElement* sink, GStreamerVideoEncoder* enc)
>     {
>@@ -532,12 +526,10 @@ public:
>         codecSpecifiInfos->codec_name = ImplementationName();
>         webrtc::CodecSpecificInfoVP8* vp8Info = &(codecSpecifiInfos->codecSpecific.VP8);
>         vp8Info->temporalIdx = 0;
>-        vp8Info->pictureId = m_pictureId;
> 
>         vp8Info->simulcastIdx = 0;
>         vp8Info->keyIdx = webrtc::kNoKeyIdx;
>         vp8Info->nonReference = GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT);
>-        vp8Info->tl0PicIdx = webrtc::kNoTl0PicIdx;
>     }
> };
> 
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>index 6991cde0aaf..d8c4069b29b 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>@@ -39,8 +39,8 @@
> #pragma clang diagnostic ignored "-Wunused-parameter"
> 
> #include <webrtc/api/peerconnectioninterface.h>
>-#include <webrtc/media/engine/webrtcvideodecoderfactory.h>
>-#include <webrtc/media/engine/webrtcvideoencoderfactory.h>
>+#include <webrtc/api/video_codecs/video_encoder_factory.h>
>+#include <webrtc/api/video_codecs/video_decoder_factory.h>
> #include <webrtc/rtc_base/scoped_ref_ptr.h>
> 
> #pragma clang diagnostic pop

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Attachments on bug 188745: 347508 | 347510 | 347512 | 347568 | 347607 | 347679 | 347698 | 347735 | 347769 | 347795 | 347808 | 347839