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tg2sip/libtgvoip/Makefile.in

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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
# Makefile.in generated by automake 1.16.1 from Makefile.am.
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# Copyright (C) 1994-2018 Free Software Foundation, Inc.
# This Makefile.in is free software; the Free Software Foundation
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# with or without modifications, as long as this notice is preserved.
# This program is distributed in the hope that it will be useful,
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@am__append_8 = -DWEBRTC_POSIX -DWEBRTC_APM_DEBUG_DUMP=0 -DWEBRTC_NS_FLOAT -I$(top_srcdir)/webrtc_dsp
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
@ENABLE_DSP_TRUE@am__append_9 = -I$(top_srcdir)/webrtc_dsp
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@am__append_10 = \
@ENABLE_DSP_TRUE@./webrtc_dsp/system_wrappers/source/field_trial.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/system_wrappers/source/metrics.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/system_wrappers/source/cpu_features.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/strings/internal/memutil.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/strings/string_view.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/strings/ascii.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/types/bad_optional_access.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/types/optional.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/base/internal/raw_logging.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/absl/base/internal/throw_delegate.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/race_checker.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/strings/string_builder.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/memory/aligned_malloc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/timeutils.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/platform_file.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/string_to_number.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/thread_checker_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/stringencode.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/stringutils.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/checks.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/platform_thread.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/logging_webrtc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/criticalsection.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/platform_thread_types.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/event.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/rtc_base/event_tracer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/third_party/rnnoise/src/rnn_vad_weights.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/third_party/rnnoise/src/kiss_fft.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/api/audio/audio_frame.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/api/audio/echo_canceller3_config.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/api/audio/echo_canceller3_factory.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/third_party/fft/fft.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filterbanks.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/transform.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_filter.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filter_functions.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lattice.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/intialize.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/entropy_coding.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_vad.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/crc.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode_bwe.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/rms_level.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_detector/moving_max.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_detector/circular_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_detector/mean_variance_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/splitting_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/gain_control_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/ns/nsx_core.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/ns/noise_suppression_x.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/ns/nsx_core_c.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/ns/ns_core.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/ns/noise_suppression.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/audio_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/typing_detection.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/include/audio_generator_factory.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/include/aec_dump.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/include/audio_processing.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/include/config.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/interpolated_gain_curve.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/agc2_common.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/gain_applier.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/limiter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/saturation_protector.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/rnn.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/features_extraction.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/fft_util.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/lp_residual.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/vector_float_frame.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/noise_level_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/fixed_digital_level_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/fixed_gain_controller.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/vad_with_level.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/limiter_db_gain_curve.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/down_sampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/signal_classifier.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/noise_spectrum_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/biquad_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/transient/moving_moments.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/transient/wpd_tree.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/transient/wpd_node.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/transient/transient_suppressor.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/transient/transient_detector.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/low_cut_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/level_estimator_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/three_band_filter_bank.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec/echo_cancellation.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec/aec_resampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec/aec_core.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/voice_detection_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_cancellation_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/gain_control_for_experimental_agc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/agc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/loudness_histogram.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/agc_manager_direct.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/legacy/analog_agc.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/legacy/digital_agc.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/agc/utility.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/audio_processing_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/audio_generator/file_audio_generator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/gain_controller2.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/residual_echo_detector.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/noise_suppression_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aecm/aecm_core.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aecm/aecm_core_c.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aecm/echo_control_mobile.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_reverb_model.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_remover_metrics.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer2.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_path_variability.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/frame_blocker.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/subtractor.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/aec3_fft.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/fullband_erle_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/suppression_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/block_processor.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/subband_erle_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller_metrics.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/vector_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/erl_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/aec_state.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/adaptive_fir_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/skew_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/block_framer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/erle_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/reverb_model.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/subtractor_output.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/subtractor_output_analyzer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/suppression_gain.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_audibility.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/moving_average.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/aec3_common.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/matched_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller2.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_remover.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/block_processor2.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/fft_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/aec3/decimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/standalone_vad.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/pitch_internal.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/vad/gmm.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/utility/ooura_fft.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/utility/delay_estimator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/window_generator.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/channel_buffer.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/fir_filter_factory.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/wav_header.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/real_fourier_ooura.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/audio_util.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/fir_filter_sse.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/push_sinc_resampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/resampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/sinc_resampler_sse.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/push_resampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/sinc_resampler.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/wav_file.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/third_party/fft4g/fft4g.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/audio_converter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/real_fourier.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/sparse_fir_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/smoothing_filter.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/fir_filter_c.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/ring_buffer.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/complex_fft.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/filter_ma_fast_q12.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/splitting_filter1.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/levinson_durbin.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/auto_corr_to_refl_coef.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/energy.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/downsample_fast.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/spl_init.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/lpc_to_refl_coef.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/cross_correlation.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/division_operations.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/auto_correlation.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/get_scaling_square.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/resample.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/min_max_operations.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/refl_coef_to_lpc.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/filter_ar.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/vector_scaling_operations.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/resample_fractional.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/real_fft.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/ilbc_specific_functions.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/randomization_functions.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/copy_set_operations.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/resample_by_2.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/get_hanning_window.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/resample_48khz.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/spl_inl.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/signal_processing/spl_sqrt.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/vad_sp.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/vad.cc \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/webrtc_vad.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/vad_filterbank.c \
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/vad_core.c \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@./webrtc_dsp/common_audio/vad/vad_gmm.c
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_TRUE@am__append_11 = -DWEBRTC_MAC
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_TRUE@am__append_12 = \
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_TRUE@webrtc_dsp/rtc_base/logging_mac.mm \
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_TRUE@webrtc_dsp/rtc_base/logging_mac.h
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_FALSE@am__append_13 = -DWEBRTC_LINUX
@ENABLE_DSP_TRUE@@TARGET_CPU_X86_TRUE@am__append_14 = \
@ENABLE_DSP_TRUE@@TARGET_CPU_X86_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_core_sse2.cc \
@ENABLE_DSP_TRUE@@TARGET_CPU_X86_TRUE@webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.cc
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@am__append_15 = -DTGVOIP_USE_CALLBACK_AUDIO_IO
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@am__append_16 = \
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@audio/AudioIOCallback.cpp
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@am__append_17 = \
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@audio/AudioIOCallback.h
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_TRUE@am__append_18 = \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/common_audio/signal_processing/complex_bit_reverse_arm.S \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@am__append_19 = -mfpu=neon -mfloat-abi=hard
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@am__append_20 = -mfpu=neon -mfloat-abi=hard
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@am__append_21 = \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/common_audio/signal_processing/cross_correlation_neon.c \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/common_audio/signal_processing/downsample_fast_neon.c \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/common_audio/signal_processing/min_max_operations_neon.c \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_core_neon.cc \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/modules/audio_processing/aecm/aecm_core_neon.cc \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/modules/audio_processing/ns/nsx_core_neon.c \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@webrtc_dsp/modules/audio_processing/utility/ooura_fft_neon.cc
# webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_FALSE@am__append_22 = \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_FALSE@webrtc_dsp/common_audio/signal_processing/complex_bit_reverse.c \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_FALSE@webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
# headers
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@am__append_23 = \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@webrtc_dsp/system_wrappers/include/field_trial.h \
@ENABLE_DSP_TRUE@webrtc_dsp/system_wrappers/include/cpu_features_wrapper.h \
@ENABLE_DSP_TRUE@webrtc_dsp/system_wrappers/include/asm_defines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/system_wrappers/include/metrics.h \
@ENABLE_DSP_TRUE@webrtc_dsp/system_wrappers/include/compile_assert_c.h \
@ENABLE_DSP_TRUE@webrtc_dsp/typedefs.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/strings/internal/memutil.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/strings/ascii.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/strings/string_view.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/types/optional.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/types/bad_optional_access.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/memory/memory.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/meta/type_traits.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/algorithm/algorithm.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/container/inlined_vector.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/policy_checks.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/port.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/config.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/invoke.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/inline_variable.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/atomic_hook.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/identity.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/raw_logging.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/internal/throw_delegate.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/attributes.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/macros.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/optimization.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/base/log_severity.h \
@ENABLE_DSP_TRUE@webrtc_dsp/absl/utility/utility.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/string_to_number.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/constructormagic.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/strings/string_builder.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/event_tracer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/stringencode.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/memory/aligned_malloc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/event.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/ignore_wundef.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/stringutils.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/arraysize.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/swap_queue.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/trace_event.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/checks.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/deprecation.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/sanitizer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/scoped_ref_ptr.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/logging.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/timeutils.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/atomicops.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/numerics/safe_minmax.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/numerics/safe_conversions.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/numerics/safe_conversions_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/numerics/safe_compare.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/unused.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/inline.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/ignore_warnings.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/asm_defines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/rtc_export.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/system/arch.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/platform_thread.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/platform_thread_types.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/protobuf_utils.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/thread_annotations.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/gtest_prod_util.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/function_view.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/criticalsection.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/refcount.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/thread_checker_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/compile_assert_c.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/type_traits.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/platform_file.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/refcounter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/thread_checker.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/race_checker.h \
@ENABLE_DSP_TRUE@webrtc_dsp/rtc_base/refcountedobject.h \
@ENABLE_DSP_TRUE@webrtc_dsp/third_party/rnnoise/src/rnn_activations.h \
@ENABLE_DSP_TRUE@webrtc_dsp/third_party/rnnoise/src/kiss_fft.h \
@ENABLE_DSP_TRUE@webrtc_dsp/third_party/rnnoise/src/rnn_vad_weights.h \
@ENABLE_DSP_TRUE@webrtc_dsp/api/audio/echo_canceller3_config.h \
@ENABLE_DSP_TRUE@webrtc_dsp/api/audio/echo_control.h \
@ENABLE_DSP_TRUE@webrtc_dsp/api/audio/audio_frame.h \
@ENABLE_DSP_TRUE@webrtc_dsp/api/audio/echo_canceller3_factory.h \
@ENABLE_DSP_TRUE@webrtc_dsp/api/array_view.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/third_party/fft/fft.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/bandwidth_info.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/include/isac.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/entropy_coding.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_vad.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/settings.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/crc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_float_type.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/codec.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/structs.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filter_functions.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_detector/moving_max.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_detector/circular_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_detector/normalized_covariance_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_detector/mean_variance_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/gain_control_for_experimental_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/rms_level.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/ns_core.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/defines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/noise_suppression.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/nsx_core.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/windows_private.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/noise_suppression_x.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/ns/nsx_defines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/residual_echo_detector.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/audio_processing_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/render_queue_item_verifier.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/audio_generator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/config.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/audio_frame_view.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/mock_audio_processing.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/gain_control.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/audio_generator_factory.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/aec_dump.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/include/audio_processing.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/interpolated_gain_curve.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/biquad_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/signal_classifier.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/vector_float_frame.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/rnn.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/test_utils.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_info.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/lp_residual.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/ring_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/features_extraction.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/fft_util.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/fixed_gain_controller.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/down_sampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/saturation_protector.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/agc2_common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/adaptive_digital_gain_applier.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/vad_with_level.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/limiter_db_gain_curve.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/fixed_digital_level_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/gain_applier.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/noise_level_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/compute_interpolated_gain_curve.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/noise_spectrum_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc2/limiter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/transient_detector.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/transient_suppressor.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/wpd_node.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/moving_moments.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/wpd_tree.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/transient/dyadic_decimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/noise_suppression_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_resampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec/echo_cancellation.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_core.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_core_optimized_methods.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec/aec_common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/voice_detection_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/legacy/analog_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/legacy/gain_control.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/legacy/digital_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/mock_agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/loudness_histogram.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/gain_map_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/utility.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/agc_manager_direct.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/agc/agc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/audio_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/splitting_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/low_cut_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/audio_generator/file_audio_generator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/three_band_filter_bank.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/echo_cancellation_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/level_estimator_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/gain_controller2.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aecm/aecm_core.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aecm/aecm_defines.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aecm/echo_control_mobile.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/subtractor_output_analyzer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/aec_state.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/suppression_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/adaptive_fir_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/matched_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/subtractor_output.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/aec3_fft.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_remover_metrics.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/subtractor.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/fft_data.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_delay_controller_metrics.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/erl_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_remover.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_path_variability.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/moving_average.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_reverb_model.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/suppression_gain.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/erle_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/subband_erle_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/block_processor.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/fullband_erle_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/skew_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/vector_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/echo_audibility.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/fft_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/aec3_common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/vector_math.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/decimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/frame_blocker.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/block_framer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/delay_estimate.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/reverb_model.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/matched_filter_lag_aggregator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/gain_control_impl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/typing_detection.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/vad_audio_proc_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/gmm.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/voice_gmm_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/noise_gmm_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/pitch_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/standalone_vad.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/delay_estimator_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/ooura_fft.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/delay_estimator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/ooura_fft_tables_common.h \
@ENABLE_DSP_TRUE@webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/mocks/mock_smoothing_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/wav_file.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/sparse_fir_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/fir_filter_sse.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/window_generator.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/ring_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/fir_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/include/audio_util.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/real_fourier_ooura.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/smoothing_filter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/resampler/sinc_resampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/resampler/include/push_resampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/resampler/include/resampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/resampler/push_sinc_resampler.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/fir_filter_factory.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/audio_converter.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/third_party/fft4g/fft4g.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/channel_buffer.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/real_fourier.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/fir_filter_neon.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/fir_filter_c.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/complex_fft_tables.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/include/signal_processing_library.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/include/real_fft.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/include/spl_inl.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/include/spl_inl_armv7.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/wav_header.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/vad_core.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/include/vad.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/include/webrtc_vad.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/vad_gmm.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/vad_sp.h \
@ENABLE_DSP_TRUE@webrtc_dsp/common_audio/vad/vad_filterbank.h
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_FALSE@am__append_24 = -DTGVOIP_NO_DSP
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
subdir = .
ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
am__aclocal_m4_deps = $(top_srcdir)/configure.ac
am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
$(ACLOCAL_M4)
DIST_COMMON = $(srcdir)/Makefile.am $(top_srcdir)/configure \
$(am__configure_deps) $(am__nobase_tgvoipinclude_HEADERS_DIST) \
$(am__DIST_COMMON)
am__CONFIG_DISTCLEAN_FILES = config.status config.cache config.log \
configure.lineno config.status.lineno
mkinstalldirs = $(install_sh) -d
CONFIG_HEADER = config.h
CONFIG_CLEAN_FILES =
CONFIG_CLEAN_VPATH_FILES =
am__vpath_adj_setup = srcdirstrip=`echo "$(srcdir)" | sed 's|.|.|g'`;
am__vpath_adj = case $$p in \
$(srcdir)/*) f=`echo "$$p" | sed "s|^$$srcdirstrip/||"`;; \
*) f=$$p;; \
esac;
am__strip_dir = f=`echo $$p | sed -e 's|^.*/||'`;
am__install_max = 40
am__nobase_strip_setup = \
srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*|]/\\\\&/g'`
am__nobase_strip = \
for p in $$list; do echo "$$p"; done | sed -e "s|$$srcdirstrip/||"
am__nobase_list = $(am__nobase_strip_setup); \
for p in $$list; do echo "$$p $$p"; done | \
sed "s| $$srcdirstrip/| |;"' / .*\//!s/ .*/ ./; s,\( .*\)/[^/]*$$,\1,' | \
$(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \
if (++n[$$2] == $(am__install_max)) \
{ print $$2, files[$$2]; n[$$2] = 0; files[$$2] = "" } } \
END { for (dir in files) print dir, files[dir] }'
am__base_list = \
sed '$$!N;$$!N;$$!N;$$!N;$$!N;$$!N;$$!N;s/\n/ /g' | \
sed '$$!N;$$!N;$$!N;$$!N;s/\n/ /g'
am__uninstall_files_from_dir = { \
test -z "$$files" \
|| { test ! -d "$$dir" && test ! -f "$$dir" && test ! -r "$$dir"; } \
|| { echo " ( cd '$$dir' && rm -f" $$files ")"; \
$(am__cd) "$$dir" && rm -f $$files; }; \
}
am__installdirs = "$(DESTDIR)$(libdir)" \
"$(DESTDIR)$(tgvoipincludedir)"
LTLIBRARIES = $(lib_LTLIBRARIES)
libtgvoip_la_LIBADD =
am__libtgvoip_la_SOURCES_DIST = VoIPController.cpp Buffers.cpp \
CongestionControl.cpp EchoCanceller.cpp JitterBuffer.cpp \
logging.cpp MediaStreamItf.cpp MessageThread.cpp \
NetworkSocket.cpp OpusDecoder.cpp OpusEncoder.cpp \
PacketReassembler.cpp VoIPGroupController.cpp \
VoIPServerConfig.cpp audio/AudioIO.cpp audio/AudioInput.cpp \
audio/AudioOutput.cpp audio/Resampler.cpp \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/posix/NetworkSocketPosix.cpp video/VideoSource.cpp \
video/VideoRenderer.cpp json11.cpp \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
os/darwin/AudioInputAudioUnit.cpp \
os/darwin/AudioOutputAudioUnit.cpp os/darwin/AudioUnitIO.cpp \
os/darwin/AudioInputAudioUnitOSX.cpp \
os/darwin/AudioOutputAudioUnitOSX.cpp \
os/darwin/DarwinSpecific.mm os/linux/AudioInputALSA.cpp \
os/linux/AudioOutputALSA.cpp os/linux/AudioOutputPulse.cpp \
os/linux/AudioInputPulse.cpp os/linux/AudioPulse.cpp \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
./webrtc_dsp/system_wrappers/source/field_trial.cc \
./webrtc_dsp/system_wrappers/source/metrics.cc \
./webrtc_dsp/system_wrappers/source/cpu_features.cc \
./webrtc_dsp/absl/strings/internal/memutil.cc \
./webrtc_dsp/absl/strings/string_view.cc \
./webrtc_dsp/absl/strings/ascii.cc \
./webrtc_dsp/absl/types/bad_optional_access.cc \
./webrtc_dsp/absl/types/optional.cc \
./webrtc_dsp/absl/base/internal/raw_logging.cc \
./webrtc_dsp/absl/base/internal/throw_delegate.cc \
./webrtc_dsp/rtc_base/race_checker.cc \
./webrtc_dsp/rtc_base/strings/string_builder.cc \
./webrtc_dsp/rtc_base/memory/aligned_malloc.cc \
./webrtc_dsp/rtc_base/timeutils.cc \
./webrtc_dsp/rtc_base/platform_file.cc \
./webrtc_dsp/rtc_base/string_to_number.cc \
./webrtc_dsp/rtc_base/thread_checker_impl.cc \
./webrtc_dsp/rtc_base/stringencode.cc \
./webrtc_dsp/rtc_base/stringutils.cc \
./webrtc_dsp/rtc_base/checks.cc \
./webrtc_dsp/rtc_base/platform_thread.cc \
./webrtc_dsp/rtc_base/logging_webrtc.cc \
./webrtc_dsp/rtc_base/criticalsection.cc \
./webrtc_dsp/rtc_base/platform_thread_types.cc \
./webrtc_dsp/rtc_base/event.cc \
./webrtc_dsp/rtc_base/event_tracer.cc \
./webrtc_dsp/third_party/rnnoise/src/rnn_vad_weights.cc \
./webrtc_dsp/third_party/rnnoise/src/kiss_fft.cc \
./webrtc_dsp/api/audio/audio_frame.cc \
./webrtc_dsp/api/audio/echo_canceller3_config.cc \
./webrtc_dsp/api/audio/echo_canceller3_factory.cc \
./webrtc_dsp/modules/third_party/fft/fft.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filterbanks.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/transform.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_filter.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filter_functions.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lattice.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/intialize.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/entropy_coding.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_vad.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/crc.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode_bwe.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac.c \
./webrtc_dsp/modules/audio_processing/rms_level.cc \
./webrtc_dsp/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc \
./webrtc_dsp/modules/audio_processing/echo_detector/moving_max.cc \
./webrtc_dsp/modules/audio_processing/echo_detector/circular_buffer.cc \
./webrtc_dsp/modules/audio_processing/echo_detector/mean_variance_estimator.cc \
./webrtc_dsp/modules/audio_processing/splitting_filter.cc \
./webrtc_dsp/modules/audio_processing/gain_control_impl.cc \
./webrtc_dsp/modules/audio_processing/ns/nsx_core.c \
./webrtc_dsp/modules/audio_processing/ns/noise_suppression_x.c \
./webrtc_dsp/modules/audio_processing/ns/nsx_core_c.c \
./webrtc_dsp/modules/audio_processing/ns/ns_core.c \
./webrtc_dsp/modules/audio_processing/ns/noise_suppression.c \
./webrtc_dsp/modules/audio_processing/audio_buffer.cc \
./webrtc_dsp/modules/audio_processing/typing_detection.cc \
./webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.cc \
./webrtc_dsp/modules/audio_processing/include/audio_generator_factory.cc \
./webrtc_dsp/modules/audio_processing/include/aec_dump.cc \
./webrtc_dsp/modules/audio_processing/include/audio_processing.cc \
./webrtc_dsp/modules/audio_processing/include/config.cc \
./webrtc_dsp/modules/audio_processing/agc2/interpolated_gain_curve.cc \
./webrtc_dsp/modules/audio_processing/agc2/agc2_common.cc \
./webrtc_dsp/modules/audio_processing/agc2/gain_applier.cc \
./webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.cc \
./webrtc_dsp/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc \
./webrtc_dsp/modules/audio_processing/agc2/limiter.cc \
./webrtc_dsp/modules/audio_processing/agc2/saturation_protector.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/rnn.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/features_extraction.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/fft_util.cc \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/lp_residual.cc \
./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc \
./webrtc_dsp/modules/audio_processing/agc2/vector_float_frame.cc \
./webrtc_dsp/modules/audio_processing/agc2/noise_level_estimator.cc \
./webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.cc \
./webrtc_dsp/modules/audio_processing/agc2/fixed_digital_level_estimator.cc \
./webrtc_dsp/modules/audio_processing/agc2/fixed_gain_controller.cc \
./webrtc_dsp/modules/audio_processing/agc2/vad_with_level.cc \
./webrtc_dsp/modules/audio_processing/agc2/limiter_db_gain_curve.cc \
./webrtc_dsp/modules/audio_processing/agc2/down_sampler.cc \
./webrtc_dsp/modules/audio_processing/agc2/signal_classifier.cc \
./webrtc_dsp/modules/audio_processing/agc2/noise_spectrum_estimator.cc \
./webrtc_dsp/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc \
./webrtc_dsp/modules/audio_processing/agc2/biquad_filter.cc \
./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc \
./webrtc_dsp/modules/audio_processing/transient/moving_moments.cc \
./webrtc_dsp/modules/audio_processing/transient/wpd_tree.cc \
./webrtc_dsp/modules/audio_processing/transient/wpd_node.cc \
./webrtc_dsp/modules/audio_processing/transient/transient_suppressor.cc \
./webrtc_dsp/modules/audio_processing/transient/transient_detector.cc \
./webrtc_dsp/modules/audio_processing/low_cut_filter.cc \
./webrtc_dsp/modules/audio_processing/level_estimator_impl.cc \
./webrtc_dsp/modules/audio_processing/three_band_filter_bank.cc \
./webrtc_dsp/modules/audio_processing/aec/echo_cancellation.cc \
./webrtc_dsp/modules/audio_processing/aec/aec_resampler.cc \
./webrtc_dsp/modules/audio_processing/aec/aec_core.cc \
./webrtc_dsp/modules/audio_processing/voice_detection_impl.cc \
./webrtc_dsp/modules/audio_processing/echo_cancellation_impl.cc \
./webrtc_dsp/modules/audio_processing/gain_control_for_experimental_agc.cc \
./webrtc_dsp/modules/audio_processing/agc/agc.cc \
./webrtc_dsp/modules/audio_processing/agc/loudness_histogram.cc \
./webrtc_dsp/modules/audio_processing/agc/agc_manager_direct.cc \
./webrtc_dsp/modules/audio_processing/agc/legacy/analog_agc.c \
./webrtc_dsp/modules/audio_processing/agc/legacy/digital_agc.c \
./webrtc_dsp/modules/audio_processing/agc/utility.cc \
./webrtc_dsp/modules/audio_processing/audio_processing_impl.cc \
./webrtc_dsp/modules/audio_processing/audio_generator/file_audio_generator.cc \
./webrtc_dsp/modules/audio_processing/gain_controller2.cc \
./webrtc_dsp/modules/audio_processing/residual_echo_detector.cc \
./webrtc_dsp/modules/audio_processing/noise_suppression_impl.cc \
./webrtc_dsp/modules/audio_processing/aecm/aecm_core.cc \
./webrtc_dsp/modules/audio_processing/aecm/aecm_core_c.cc \
./webrtc_dsp/modules/audio_processing/aecm/echo_control_mobile.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_reverb_model.cc \
./webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_remover_metrics.cc \
./webrtc_dsp/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer2.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_path_variability.cc \
./webrtc_dsp/modules/audio_processing/aec3/frame_blocker.cc \
./webrtc_dsp/modules/audio_processing/aec3/subtractor.cc \
./webrtc_dsp/modules/audio_processing/aec3/aec3_fft.cc \
./webrtc_dsp/modules/audio_processing/aec3/fullband_erle_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/suppression_filter.cc \
./webrtc_dsp/modules/audio_processing/aec3/block_processor.cc \
./webrtc_dsp/modules/audio_processing/aec3/subband_erle_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller_metrics.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/vector_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/erl_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/aec_state.cc \
./webrtc_dsp/modules/audio_processing/aec3/adaptive_fir_filter.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.cc \
./webrtc_dsp/modules/audio_processing/aec3/skew_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/block_framer.cc \
./webrtc_dsp/modules/audio_processing/aec3/erle_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/reverb_model.cc \
./webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/subtractor_output.cc \
./webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.cc \
./webrtc_dsp/modules/audio_processing/aec3/subtractor_output_analyzer.cc \
./webrtc_dsp/modules/audio_processing/aec3/suppression_gain.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_audibility.cc \
./webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.cc \
./webrtc_dsp/modules/audio_processing/aec3/moving_average.cc \
./webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/aec3_common.cc \
./webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/matched_filter.cc \
./webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.cc \
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller2.cc \
./webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.cc \
./webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_remover.cc \
./webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/block_processor2.cc \
./webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.cc \
./webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/fft_buffer.cc \
./webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.cc \
./webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.cc \
./webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.cc \
./webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.cc \
./webrtc_dsp/modules/audio_processing/aec3/decimator.cc \
./webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.cc \
./webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.cc \
./webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.cc \
./webrtc_dsp/modules/audio_processing/vad/standalone_vad.cc \
./webrtc_dsp/modules/audio_processing/vad/pitch_internal.cc \
./webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.cc \
./webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.cc \
./webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.cc \
./webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.cc \
./webrtc_dsp/modules/audio_processing/vad/gmm.cc \
./webrtc_dsp/modules/audio_processing/utility/ooura_fft.cc \
./webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.cc \
./webrtc_dsp/modules/audio_processing/utility/delay_estimator.cc \
./webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.cc \
./webrtc_dsp/common_audio/window_generator.cc \
./webrtc_dsp/common_audio/channel_buffer.cc \
./webrtc_dsp/common_audio/fir_filter_factory.cc \
./webrtc_dsp/common_audio/wav_header.cc \
./webrtc_dsp/common_audio/real_fourier_ooura.cc \
./webrtc_dsp/common_audio/audio_util.cc \
./webrtc_dsp/common_audio/fir_filter_sse.cc \
./webrtc_dsp/common_audio/resampler/push_sinc_resampler.cc \
./webrtc_dsp/common_audio/resampler/resampler.cc \
./webrtc_dsp/common_audio/resampler/sinc_resampler_sse.cc \
./webrtc_dsp/common_audio/resampler/push_resampler.cc \
./webrtc_dsp/common_audio/resampler/sinc_resampler.cc \
./webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.cc \
./webrtc_dsp/common_audio/wav_file.cc \
./webrtc_dsp/common_audio/third_party/fft4g/fft4g.c \
./webrtc_dsp/common_audio/audio_converter.cc \
./webrtc_dsp/common_audio/real_fourier.cc \
./webrtc_dsp/common_audio/sparse_fir_filter.cc \
./webrtc_dsp/common_audio/smoothing_filter.cc \
./webrtc_dsp/common_audio/fir_filter_c.cc \
./webrtc_dsp/common_audio/ring_buffer.c \
./webrtc_dsp/common_audio/signal_processing/complex_fft.c \
./webrtc_dsp/common_audio/signal_processing/filter_ma_fast_q12.c \
./webrtc_dsp/common_audio/signal_processing/splitting_filter1.c \
./webrtc_dsp/common_audio/signal_processing/levinson_durbin.c \
./webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.cc \
./webrtc_dsp/common_audio/signal_processing/auto_corr_to_refl_coef.c \
./webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.c \
./webrtc_dsp/common_audio/signal_processing/energy.c \
./webrtc_dsp/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c \
./webrtc_dsp/common_audio/signal_processing/downsample_fast.c \
./webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12.c \
./webrtc_dsp/common_audio/signal_processing/spl_init.c \
./webrtc_dsp/common_audio/signal_processing/lpc_to_refl_coef.c \
./webrtc_dsp/common_audio/signal_processing/cross_correlation.c \
./webrtc_dsp/common_audio/signal_processing/division_operations.c \
./webrtc_dsp/common_audio/signal_processing/auto_correlation.c \
./webrtc_dsp/common_audio/signal_processing/get_scaling_square.c \
./webrtc_dsp/common_audio/signal_processing/resample.c \
./webrtc_dsp/common_audio/signal_processing/min_max_operations.c \
./webrtc_dsp/common_audio/signal_processing/refl_coef_to_lpc.c \
./webrtc_dsp/common_audio/signal_processing/filter_ar.c \
./webrtc_dsp/common_audio/signal_processing/vector_scaling_operations.c \
./webrtc_dsp/common_audio/signal_processing/resample_fractional.c \
./webrtc_dsp/common_audio/signal_processing/real_fft.c \
./webrtc_dsp/common_audio/signal_processing/ilbc_specific_functions.c \
./webrtc_dsp/common_audio/signal_processing/randomization_functions.c \
./webrtc_dsp/common_audio/signal_processing/copy_set_operations.c \
./webrtc_dsp/common_audio/signal_processing/resample_by_2.c \
./webrtc_dsp/common_audio/signal_processing/get_hanning_window.c \
./webrtc_dsp/common_audio/signal_processing/resample_48khz.c \
./webrtc_dsp/common_audio/signal_processing/spl_inl.c \
./webrtc_dsp/common_audio/signal_processing/spl_sqrt.c \
./webrtc_dsp/common_audio/vad/vad_sp.c \
./webrtc_dsp/common_audio/vad/vad.cc \
./webrtc_dsp/common_audio/vad/webrtc_vad.c \
./webrtc_dsp/common_audio/vad/vad_filterbank.c \
./webrtc_dsp/common_audio/vad/vad_core.c \
./webrtc_dsp/common_audio/vad/vad_gmm.c \
webrtc_dsp/rtc_base/logging_mac.mm \
webrtc_dsp/rtc_base/logging_mac.h \
webrtc_dsp/modules/audio_processing/aec/aec_core_sse2.cc \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.cc \
audio/AudioIOCallback.cpp \
webrtc_dsp/common_audio/signal_processing/complex_bit_reverse_arm.S \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
webrtc_dsp/common_audio/signal_processing/cross_correlation_neon.c \
webrtc_dsp/common_audio/signal_processing/downsample_fast_neon.c \
webrtc_dsp/common_audio/signal_processing/min_max_operations_neon.c \
webrtc_dsp/modules/audio_processing/aec/aec_core_neon.cc \
webrtc_dsp/modules/audio_processing/aecm/aecm_core_neon.cc \
webrtc_dsp/modules/audio_processing/ns/nsx_core_neon.c \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_neon.cc \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
webrtc_dsp/common_audio/signal_processing/complex_bit_reverse.c \
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
webrtc_dsp/system_wrappers/include/field_trial.h \
webrtc_dsp/system_wrappers/include/cpu_features_wrapper.h \
webrtc_dsp/system_wrappers/include/asm_defines.h \
webrtc_dsp/system_wrappers/include/metrics.h \
webrtc_dsp/system_wrappers/include/compile_assert_c.h \
webrtc_dsp/typedefs.h \
webrtc_dsp/absl/strings/internal/memutil.h \
webrtc_dsp/absl/strings/ascii.h \
webrtc_dsp/absl/strings/string_view.h \
webrtc_dsp/absl/types/optional.h \
webrtc_dsp/absl/types/bad_optional_access.h \
webrtc_dsp/absl/memory/memory.h \
webrtc_dsp/absl/meta/type_traits.h \
webrtc_dsp/absl/algorithm/algorithm.h \
webrtc_dsp/absl/container/inlined_vector.h \
webrtc_dsp/absl/base/policy_checks.h \
webrtc_dsp/absl/base/port.h webrtc_dsp/absl/base/config.h \
webrtc_dsp/absl/base/internal/invoke.h \
webrtc_dsp/absl/base/internal/inline_variable.h \
webrtc_dsp/absl/base/internal/atomic_hook.h \
webrtc_dsp/absl/base/internal/identity.h \
webrtc_dsp/absl/base/internal/raw_logging.h \
webrtc_dsp/absl/base/internal/throw_delegate.h \
webrtc_dsp/absl/base/attributes.h \
webrtc_dsp/absl/base/macros.h \
webrtc_dsp/absl/base/optimization.h \
webrtc_dsp/absl/base/log_severity.h \
webrtc_dsp/absl/utility/utility.h \
webrtc_dsp/rtc_base/string_to_number.h \
webrtc_dsp/rtc_base/constructormagic.h \
webrtc_dsp/rtc_base/strings/string_builder.h \
webrtc_dsp/rtc_base/event_tracer.h \
webrtc_dsp/rtc_base/stringencode.h \
webrtc_dsp/rtc_base/memory/aligned_malloc.h \
webrtc_dsp/rtc_base/event.h \
webrtc_dsp/rtc_base/ignore_wundef.h \
webrtc_dsp/rtc_base/stringutils.h \
webrtc_dsp/rtc_base/arraysize.h \
webrtc_dsp/rtc_base/swap_queue.h \
webrtc_dsp/rtc_base/trace_event.h webrtc_dsp/rtc_base/checks.h \
webrtc_dsp/rtc_base/deprecation.h \
webrtc_dsp/rtc_base/sanitizer.h \
webrtc_dsp/rtc_base/scoped_ref_ptr.h \
webrtc_dsp/rtc_base/logging.h webrtc_dsp/rtc_base/timeutils.h \
webrtc_dsp/rtc_base/atomicops.h \
webrtc_dsp/rtc_base/numerics/safe_minmax.h \
webrtc_dsp/rtc_base/numerics/safe_conversions.h \
webrtc_dsp/rtc_base/numerics/safe_conversions_impl.h \
webrtc_dsp/rtc_base/numerics/safe_compare.h \
webrtc_dsp/rtc_base/system/unused.h \
webrtc_dsp/rtc_base/system/inline.h \
webrtc_dsp/rtc_base/system/ignore_warnings.h \
webrtc_dsp/rtc_base/system/asm_defines.h \
webrtc_dsp/rtc_base/system/rtc_export.h \
webrtc_dsp/rtc_base/system/arch.h \
webrtc_dsp/rtc_base/platform_thread.h \
webrtc_dsp/rtc_base/platform_thread_types.h \
webrtc_dsp/rtc_base/protobuf_utils.h \
webrtc_dsp/rtc_base/thread_annotations.h \
webrtc_dsp/rtc_base/gtest_prod_util.h \
webrtc_dsp/rtc_base/function_view.h \
webrtc_dsp/rtc_base/criticalsection.h \
webrtc_dsp/rtc_base/refcount.h \
webrtc_dsp/rtc_base/thread_checker_impl.h \
webrtc_dsp/rtc_base/compile_assert_c.h \
webrtc_dsp/rtc_base/type_traits.h \
webrtc_dsp/rtc_base/platform_file.h \
webrtc_dsp/rtc_base/refcounter.h \
webrtc_dsp/rtc_base/thread_checker.h \
webrtc_dsp/rtc_base/race_checker.h \
webrtc_dsp/rtc_base/refcountedobject.h \
webrtc_dsp/third_party/rnnoise/src/rnn_activations.h \
webrtc_dsp/third_party/rnnoise/src/kiss_fft.h \
webrtc_dsp/third_party/rnnoise/src/rnn_vad_weights.h \
webrtc_dsp/api/audio/echo_canceller3_config.h \
webrtc_dsp/api/audio/echo_control.h \
webrtc_dsp/api/audio/audio_frame.h \
webrtc_dsp/api/audio/echo_canceller3_factory.h \
webrtc_dsp/api/array_view.h \
webrtc_dsp/modules/third_party/fft/fft.h \
webrtc_dsp/modules/audio_coding/codecs/isac/bandwidth_info.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/include/isac.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/entropy_coding.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_vad.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/settings.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/crc.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_float_type.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/codec.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/structs.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filter_functions.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_filter.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_tables.h \
webrtc_dsp/modules/audio_processing/echo_detector/moving_max.h \
webrtc_dsp/modules/audio_processing/echo_detector/circular_buffer.h \
webrtc_dsp/modules/audio_processing/echo_detector/normalized_covariance_estimator.h \
webrtc_dsp/modules/audio_processing/echo_detector/mean_variance_estimator.h \
webrtc_dsp/modules/audio_processing/gain_control_for_experimental_agc.h \
webrtc_dsp/modules/audio_processing/rms_level.h \
webrtc_dsp/modules/audio_processing/ns/ns_core.h \
webrtc_dsp/modules/audio_processing/ns/defines.h \
webrtc_dsp/modules/audio_processing/ns/noise_suppression.h \
webrtc_dsp/modules/audio_processing/ns/nsx_core.h \
webrtc_dsp/modules/audio_processing/ns/windows_private.h \
webrtc_dsp/modules/audio_processing/ns/noise_suppression_x.h \
webrtc_dsp/modules/audio_processing/ns/nsx_defines.h \
webrtc_dsp/modules/audio_processing/residual_echo_detector.h \
webrtc_dsp/modules/audio_processing/audio_processing_impl.h \
webrtc_dsp/modules/audio_processing/render_queue_item_verifier.h \
webrtc_dsp/modules/audio_processing/include/audio_generator.h \
webrtc_dsp/modules/audio_processing/include/config.h \
webrtc_dsp/modules/audio_processing/include/audio_frame_view.h \
webrtc_dsp/modules/audio_processing/include/mock_audio_processing.h \
webrtc_dsp/modules/audio_processing/include/gain_control.h \
webrtc_dsp/modules/audio_processing/include/audio_generator_factory.h \
webrtc_dsp/modules/audio_processing/include/aec_dump.h \
webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.h \
webrtc_dsp/modules/audio_processing/include/audio_processing.h \
webrtc_dsp/modules/audio_processing/agc2/interpolated_gain_curve.h \
webrtc_dsp/modules/audio_processing/agc2/biquad_filter.h \
webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.h \
webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator.h \
webrtc_dsp/modules/audio_processing/agc2/signal_classifier.h \
webrtc_dsp/modules/audio_processing/agc2/vector_float_frame.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/rnn.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/test_utils.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_info.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/lp_residual.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/ring_buffer.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/features_extraction.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/common.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/fft_util.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search.h \
webrtc_dsp/modules/audio_processing/agc2/fixed_gain_controller.h \
webrtc_dsp/modules/audio_processing/agc2/down_sampler.h \
webrtc_dsp/modules/audio_processing/agc2/saturation_protector.h \
webrtc_dsp/modules/audio_processing/agc2/agc2_common.h \
webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h \
webrtc_dsp/modules/audio_processing/agc2/adaptive_digital_gain_applier.h \
webrtc_dsp/modules/audio_processing/agc2/vad_with_level.h \
webrtc_dsp/modules/audio_processing/agc2/limiter_db_gain_curve.h \
webrtc_dsp/modules/audio_processing/agc2/fixed_digital_level_estimator.h \
webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.h \
webrtc_dsp/modules/audio_processing/agc2/gain_applier.h \
webrtc_dsp/modules/audio_processing/agc2/noise_level_estimator.h \
webrtc_dsp/modules/audio_processing/agc2/compute_interpolated_gain_curve.h \
webrtc_dsp/modules/audio_processing/agc2/noise_spectrum_estimator.h \
webrtc_dsp/modules/audio_processing/agc2/limiter.h \
webrtc_dsp/modules/audio_processing/transient/transient_detector.h \
webrtc_dsp/modules/audio_processing/transient/transient_suppressor.h \
webrtc_dsp/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h \
webrtc_dsp/modules/audio_processing/transient/common.h \
webrtc_dsp/modules/audio_processing/transient/wpd_node.h \
webrtc_dsp/modules/audio_processing/transient/moving_moments.h \
webrtc_dsp/modules/audio_processing/transient/wpd_tree.h \
webrtc_dsp/modules/audio_processing/transient/dyadic_decimator.h \
webrtc_dsp/modules/audio_processing/noise_suppression_impl.h \
webrtc_dsp/modules/audio_processing/aec/aec_resampler.h \
webrtc_dsp/modules/audio_processing/aec/echo_cancellation.h \
webrtc_dsp/modules/audio_processing/aec/aec_core.h \
webrtc_dsp/modules/audio_processing/aec/aec_core_optimized_methods.h \
webrtc_dsp/modules/audio_processing/aec/aec_common.h \
webrtc_dsp/modules/audio_processing/voice_detection_impl.h \
webrtc_dsp/modules/audio_processing/agc/legacy/analog_agc.h \
webrtc_dsp/modules/audio_processing/agc/legacy/gain_control.h \
webrtc_dsp/modules/audio_processing/agc/legacy/digital_agc.h \
webrtc_dsp/modules/audio_processing/agc/mock_agc.h \
webrtc_dsp/modules/audio_processing/agc/loudness_histogram.h \
webrtc_dsp/modules/audio_processing/agc/gain_map_internal.h \
webrtc_dsp/modules/audio_processing/agc/utility.h \
webrtc_dsp/modules/audio_processing/agc/agc_manager_direct.h \
webrtc_dsp/modules/audio_processing/agc/agc.h \
webrtc_dsp/modules/audio_processing/common.h \
webrtc_dsp/modules/audio_processing/audio_buffer.h \
webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.h \
webrtc_dsp/modules/audio_processing/splitting_filter.h \
webrtc_dsp/modules/audio_processing/low_cut_filter.h \
webrtc_dsp/modules/audio_processing/audio_generator/file_audio_generator.h \
webrtc_dsp/modules/audio_processing/three_band_filter_bank.h \
webrtc_dsp/modules/audio_processing/echo_cancellation_impl.h \
webrtc_dsp/modules/audio_processing/level_estimator_impl.h \
webrtc_dsp/modules/audio_processing/gain_controller2.h \
webrtc_dsp/modules/audio_processing/aecm/aecm_core.h \
webrtc_dsp/modules/audio_processing/aecm/aecm_defines.h \
webrtc_dsp/modules/audio_processing/aecm/echo_control_mobile.h \
webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/subtractor_output_analyzer.h \
webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.h \
webrtc_dsp/modules/audio_processing/aec3/aec_state.h \
webrtc_dsp/modules/audio_processing/aec3/suppression_filter.h \
webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/adaptive_fir_filter.h \
webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.h \
webrtc_dsp/modules/audio_processing/aec3/matched_filter.h \
webrtc_dsp/modules/audio_processing/aec3/subtractor_output.h \
webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.h \
webrtc_dsp/modules/audio_processing/aec3/aec3_fft.h \
webrtc_dsp/modules/audio_processing/aec3/echo_remover_metrics.h \
webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.h \
webrtc_dsp/modules/audio_processing/aec3/subtractor.h \
webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.h \
webrtc_dsp/modules/audio_processing/aec3/fft_data.h \
webrtc_dsp/modules/audio_processing/aec3/render_delay_controller_metrics.h \
webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.h \
webrtc_dsp/modules/audio_processing/aec3/erl_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/echo_remover.h \
webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/echo_path_variability.h \
webrtc_dsp/modules/audio_processing/aec3/moving_average.h \
webrtc_dsp/modules/audio_processing/aec3/render_reverb_model.h \
webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.h \
webrtc_dsp/modules/audio_processing/aec3/suppression_gain.h \
webrtc_dsp/modules/audio_processing/aec3/erle_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/subband_erle_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/block_processor.h \
webrtc_dsp/modules/audio_processing/aec3/fullband_erle_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.h \
webrtc_dsp/modules/audio_processing/aec3/skew_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/render_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.h \
webrtc_dsp/modules/audio_processing/aec3/vector_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.h \
webrtc_dsp/modules/audio_processing/aec3/echo_audibility.h \
webrtc_dsp/modules/audio_processing/aec3/fft_buffer.h \
webrtc_dsp/modules/audio_processing/aec3/aec3_common.h \
webrtc_dsp/modules/audio_processing/aec3/vector_math.h \
webrtc_dsp/modules/audio_processing/aec3/decimator.h \
webrtc_dsp/modules/audio_processing/aec3/frame_blocker.h \
webrtc_dsp/modules/audio_processing/aec3/block_framer.h \
webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.h \
webrtc_dsp/modules/audio_processing/aec3/delay_estimate.h \
webrtc_dsp/modules/audio_processing/aec3/reverb_model.h \
webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.h \
webrtc_dsp/modules/audio_processing/aec3/matched_filter_lag_aggregator.h \
webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.h \
webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.h \
webrtc_dsp/modules/audio_processing/gain_control_impl.h \
webrtc_dsp/modules/audio_processing/typing_detection.h \
webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.h \
webrtc_dsp/modules/audio_processing/vad/vad_audio_proc_internal.h \
webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.h \
webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.h \
webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.h \
webrtc_dsp/modules/audio_processing/vad/gmm.h \
webrtc_dsp/modules/audio_processing/vad/common.h \
webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.h \
webrtc_dsp/modules/audio_processing/vad/voice_gmm_tables.h \
webrtc_dsp/modules/audio_processing/vad/noise_gmm_tables.h \
webrtc_dsp/modules/audio_processing/vad/pitch_internal.h \
webrtc_dsp/modules/audio_processing/vad/standalone_vad.h \
webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.h \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h \
webrtc_dsp/modules/audio_processing/utility/delay_estimator_internal.h \
webrtc_dsp/modules/audio_processing/utility/ooura_fft.h \
webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.h \
webrtc_dsp/modules/audio_processing/utility/delay_estimator.h \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_tables_common.h \
webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.h \
webrtc_dsp/common_audio/mocks/mock_smoothing_filter.h \
webrtc_dsp/common_audio/wav_file.h \
webrtc_dsp/common_audio/sparse_fir_filter.h \
webrtc_dsp/common_audio/fir_filter_sse.h \
webrtc_dsp/common_audio/window_generator.h \
webrtc_dsp/common_audio/ring_buffer.h \
webrtc_dsp/common_audio/fir_filter.h \
webrtc_dsp/common_audio/include/audio_util.h \
webrtc_dsp/common_audio/real_fourier_ooura.h \
webrtc_dsp/common_audio/smoothing_filter.h \
webrtc_dsp/common_audio/resampler/sinc_resampler.h \
webrtc_dsp/common_audio/resampler/include/push_resampler.h \
webrtc_dsp/common_audio/resampler/include/resampler.h \
webrtc_dsp/common_audio/resampler/push_sinc_resampler.h \
webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.h \
webrtc_dsp/common_audio/fir_filter_factory.h \
webrtc_dsp/common_audio/audio_converter.h \
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h \
webrtc_dsp/common_audio/third_party/fft4g/fft4g.h \
webrtc_dsp/common_audio/channel_buffer.h \
webrtc_dsp/common_audio/real_fourier.h \
webrtc_dsp/common_audio/fir_filter_neon.h \
webrtc_dsp/common_audio/fir_filter_c.h \
webrtc_dsp/common_audio/signal_processing/complex_fft_tables.h \
webrtc_dsp/common_audio/signal_processing/include/signal_processing_library.h \
webrtc_dsp/common_audio/signal_processing/include/real_fft.h \
webrtc_dsp/common_audio/signal_processing/include/spl_inl.h \
webrtc_dsp/common_audio/signal_processing/include/spl_inl_armv7.h \
webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.h \
webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.h \
webrtc_dsp/common_audio/wav_header.h \
webrtc_dsp/common_audio/vad/vad_core.h \
webrtc_dsp/common_audio/vad/include/vad.h \
webrtc_dsp/common_audio/vad/include/webrtc_vad.h \
webrtc_dsp/common_audio/vad/vad_gmm.h \
webrtc_dsp/common_audio/vad/vad_sp.h \
webrtc_dsp/common_audio/vad/vad_filterbank.h VoIPController.h \
Buffers.h BlockingQueue.h PrivateDefines.h CongestionControl.h \
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EchoCanceller.h JitterBuffer.h logging.h threading.h \
MediaStreamItf.h MessageThread.h NetworkSocket.h OpusDecoder.h \
OpusEncoder.h PacketReassembler.h VoIPServerConfig.h \
audio/AudioIO.h audio/AudioInput.h audio/AudioOutput.h \
audio/Resampler.h os/posix/NetworkSocketPosix.h \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
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video/VideoSource.h video/VideoRenderer.h json11.hpp utils.h \
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2018-10-03 14:34:43 +00:00
os/darwin/AudioInputAudioUnit.h \
os/darwin/AudioOutputAudioUnit.h os/darwin/AudioUnitIO.h \
os/darwin/AudioInputAudioUnitOSX.h \
os/darwin/AudioOutputAudioUnitOSX.h os/darwin/DarwinSpecific.h \
os/linux/AudioInputALSA.h os/linux/AudioOutputALSA.h \
os/linux/AudioOutputPulse.h os/linux/AudioInputPulse.h \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/linux/AudioPulse.h os/linux/PulseFunctions.h \
audio/AudioIOCallback.h
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/splitting_filter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/gain_control_impl.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/ns/nsx_core.lo \
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@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/ns/nsx_core_c.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/ns/ns_core.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/ns/noise_suppression.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/audio_buffer.lo \
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@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/skew_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/block_framer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/erle_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/reverb_model.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.lo \
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@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/subtractor_output.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.lo \
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@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/suppression_gain.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/echo_audibility.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/moving_average.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/aec3_common.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/matched_filter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller2.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/echo_remover.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/block_processor2.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/fft_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/aec3/decimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/standalone_vad.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/pitch_internal.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/vad/gmm.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/utility/ooura_fft.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/utility/delay_estimator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/window_generator.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/channel_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/fir_filter_factory.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/wav_header.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/real_fourier_ooura.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/audio_util.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/fir_filter_sse.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/push_sinc_resampler.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/resampler.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/sinc_resampler_sse.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/push_resampler.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/sinc_resampler.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/wav_file.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/third_party/fft4g/fft4g.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/audio_converter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/real_fourier.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/sparse_fir_filter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/smoothing_filter.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/fir_filter_c.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/ring_buffer.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/complex_fft.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/filter_ma_fast_q12.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/splitting_filter1.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/levinson_durbin.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/auto_corr_to_refl_coef.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/energy.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/sqrt_of_one_minus_x_squared.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/downsample_fast.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/spl_init.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/lpc_to_refl_coef.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/cross_correlation.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/division_operations.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/auto_correlation.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/get_scaling_square.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/resample.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/min_max_operations.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/refl_coef_to_lpc.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/filter_ar.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/vector_scaling_operations.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/resample_fractional.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/real_fft.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/ilbc_specific_functions.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/randomization_functions.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/copy_set_operations.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/resample_by_2.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/get_hanning_window.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/resample_48khz.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/spl_inl.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/signal_processing/spl_sqrt.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/vad_sp.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/vad.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/webrtc_vad.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/vad_filterbank.lo \
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/vad_core.lo \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@ ./webrtc_dsp/common_audio/vad/vad_gmm.lo
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@@TARGET_OS_OSX_TRUE@am__objects_5 = webrtc_dsp/rtc_base/logging_mac.lo
@ENABLE_DSP_TRUE@@TARGET_CPU_X86_TRUE@am__objects_6 = webrtc_dsp/modules/audio_processing/aec/aec_core_sse2.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_X86_TRUE@ webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.lo
@ENABLE_AUDIO_CALLBACK_TRUE@@ENABLE_DSP_TRUE@am__objects_7 = audio/AudioIOCallback.lo
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_TRUE@am__objects_8 = webrtc_dsp/common_audio/signal_processing/complex_bit_reverse_arm.lo \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.lo
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@am__objects_9 = webrtc_dsp/common_audio/signal_processing/cross_correlation_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/common_audio/signal_processing/downsample_fast_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/common_audio/signal_processing/min_max_operations_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/modules/audio_processing/aec/aec_core_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/modules/audio_processing/aecm/aecm_core_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/modules/audio_processing/ns/nsx_core_neon.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARMV7_TRUE@@TARGET_CPU_ARM_TRUE@ webrtc_dsp/modules/audio_processing/utility/ooura_fft_neon.lo
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_FALSE@am__objects_10 = webrtc_dsp/common_audio/signal_processing/complex_bit_reverse.lo \
@ENABLE_DSP_TRUE@@TARGET_CPU_ARM_FALSE@ webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.lo
am__objects_11 =
am__objects_12 = VoIPController.lo Buffers.lo CongestionControl.lo \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
EchoCanceller.lo JitterBuffer.lo logging.lo MediaStreamItf.lo \
MessageThread.lo NetworkSocket.lo OpusDecoder.lo \
OpusEncoder.lo PacketReassembler.lo VoIPGroupController.lo \
VoIPServerConfig.lo audio/AudioIO.lo audio/AudioInput.lo \
audio/AudioOutput.lo audio/Resampler.lo \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/posix/NetworkSocketPosix.lo video/VideoSource.lo \
video/VideoRenderer.lo json11.lo $(am__objects_1) \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
$(am__objects_2) $(am__objects_3) $(am__objects_4) \
$(am__objects_5) $(am__objects_6) $(am__objects_7) \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
$(am__objects_8) $(am__objects_9) $(am__objects_10) \
$(am__objects_11)
am__objects_13 = $(am__objects_11) $(am__objects_11) $(am__objects_11) \
$(am__objects_11)
am_libtgvoip_la_OBJECTS = $(am__objects_12) $(am__objects_13)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
libtgvoip_la_OBJECTS = $(am_libtgvoip_la_OBJECTS)
AM_V_lt = $(am__v_lt_@AM_V@)
am__v_lt_ = $(am__v_lt_@AM_DEFAULT_V@)
am__v_lt_0 = --silent
am__v_lt_1 =
AM_V_P = $(am__v_P_@AM_V@)
am__v_P_ = $(am__v_P_@AM_DEFAULT_V@)
am__v_P_0 = false
am__v_P_1 = :
AM_V_GEN = $(am__v_GEN_@AM_V@)
am__v_GEN_ = $(am__v_GEN_@AM_DEFAULT_V@)
am__v_GEN_0 = @echo " GEN " $@;
am__v_GEN_1 =
AM_V_at = $(am__v_at_@AM_V@)
am__v_at_ = $(am__v_at_@AM_DEFAULT_V@)
am__v_at_0 = @
am__v_at_1 =
DEFAULT_INCLUDES = -I.@am__isrc@
depcomp = $(SHELL) $(top_srcdir)/depcomp
am__maybe_remake_depfiles = depfiles
am__depfiles_remade = ./$(DEPDIR)/Buffers.Plo \
./$(DEPDIR)/CongestionControl.Plo \
./$(DEPDIR)/EchoCanceller.Plo ./$(DEPDIR)/JitterBuffer.Plo \
./$(DEPDIR)/MediaStreamItf.Plo ./$(DEPDIR)/MessageThread.Plo \
./$(DEPDIR)/NetworkSocket.Plo ./$(DEPDIR)/OpusDecoder.Plo \
./$(DEPDIR)/OpusEncoder.Plo ./$(DEPDIR)/PacketReassembler.Plo \
./$(DEPDIR)/VoIPController.Plo \
./$(DEPDIR)/VoIPGroupController.Plo \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
./$(DEPDIR)/VoIPServerConfig.Plo ./$(DEPDIR)/json11.Plo \
./$(DEPDIR)/logging.Plo \
./webrtc_dsp/absl/base/internal/$(DEPDIR)/raw_logging.Plo \
./webrtc_dsp/absl/base/internal/$(DEPDIR)/throw_delegate.Plo \
./webrtc_dsp/absl/strings/$(DEPDIR)/ascii.Plo \
./webrtc_dsp/absl/strings/$(DEPDIR)/string_view.Plo \
./webrtc_dsp/absl/strings/internal/$(DEPDIR)/memutil.Plo \
./webrtc_dsp/absl/types/$(DEPDIR)/bad_optional_access.Plo \
./webrtc_dsp/absl/types/$(DEPDIR)/optional.Plo \
./webrtc_dsp/api/audio/$(DEPDIR)/audio_frame.Plo \
./webrtc_dsp/api/audio/$(DEPDIR)/echo_canceller3_config.Plo \
./webrtc_dsp/api/audio/$(DEPDIR)/echo_canceller3_factory.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/audio_converter.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/audio_util.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/channel_buffer.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/fir_filter_c.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/fir_filter_factory.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/fir_filter_sse.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/real_fourier.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/real_fourier_ooura.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/ring_buffer.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/smoothing_filter.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/sparse_fir_filter.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/wav_file.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/wav_header.Plo \
./webrtc_dsp/common_audio/$(DEPDIR)/window_generator.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/push_resampler.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/push_sinc_resampler.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/resampler.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/sinc_resampler.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/sinc_resampler_sse.Plo \
./webrtc_dsp/common_audio/resampler/$(DEPDIR)/sinusoidal_linear_chirp_source.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/auto_corr_to_refl_coef.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/auto_correlation.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/complex_fft.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/copy_set_operations.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/cross_correlation.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/division_operations.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/dot_product_with_scale.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/downsample_fast.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/energy.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/filter_ar.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/filter_ar_fast_q12.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/filter_ma_fast_q12.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/get_hanning_window.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/get_scaling_square.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/ilbc_specific_functions.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/levinson_durbin.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/lpc_to_refl_coef.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/min_max_operations.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/randomization_functions.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/real_fft.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/refl_coef_to_lpc.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/resample.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/resample_48khz.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/resample_by_2.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/resample_by_2_internal.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/resample_fractional.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/spl_init.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/spl_inl.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/spl_sqrt.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/splitting_filter1.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/sqrt_of_one_minus_x_squared.Plo \
./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/vector_scaling_operations.Plo \
./webrtc_dsp/common_audio/third_party/fft4g/$(DEPDIR)/fft4g.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/vad.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/vad_core.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/vad_filterbank.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/vad_gmm.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/vad_sp.Plo \
./webrtc_dsp/common_audio/vad/$(DEPDIR)/webrtc_vad.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/arith_routines.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/arith_routines_hist.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/arith_routines_logist.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/bandwidth_estimator.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/crc.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/decode.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/decode_bwe.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/encode.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/encode_lpc_swb.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/entropy_coding.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/filter_functions.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/filterbanks.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/intialize.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/isac.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/isac_vad.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lattice.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lpc_analysis.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lpc_gain_swb_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lpc_shape_swb12_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lpc_shape_swb16_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/lpc_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/pitch_estimator.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/pitch_filter.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/pitch_gain_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/pitch_lag_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/spectrum_ar_model_tables.Plo \
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/transform.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/audio_buffer.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/audio_processing_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/echo_cancellation_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/echo_control_mobile_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/gain_control_for_experimental_agc.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/gain_control_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/gain_controller2.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/level_estimator_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/low_cut_filter.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/noise_suppression_impl.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/residual_echo_detector.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/rms_level.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/splitting_filter.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/three_band_filter_bank.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/typing_detection.Plo \
./webrtc_dsp/modules/audio_processing/$(DEPDIR)/voice_detection_impl.Plo \
./webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/aec_core.Plo \
./webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/aec_resampler.Plo \
./webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/echo_cancellation.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/adaptive_fir_filter.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/aec3_common.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/aec3_fft.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/aec_state.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/block_delay_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/block_framer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/block_processor.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/block_processor2.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/block_processor_metrics.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/cascaded_biquad_filter.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/comfort_noise_generator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/decimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/downsampled_render_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_audibility.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_canceller3.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_path_delay_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_path_variability.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_remover.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/echo_remover_metrics.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/erl_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/erle_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/fft_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/filter_analyzer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/frame_blocker.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/fullband_erle_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/main_filter_update_gain.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/matched_filter.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/matched_filter_lag_aggregator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/matrix_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/moving_average.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_delay_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_delay_buffer2.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_delay_controller.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_delay_controller2.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_delay_controller_metrics.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_reverb_model.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/render_signal_analyzer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/residual_echo_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/reverb_decay_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/reverb_frequency_response.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/reverb_model.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/reverb_model_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/reverb_model_fallback.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/shadow_filter_update_gain.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/skew_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/stationarity_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/subband_erle_estimator.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/subtractor.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/subtractor_output.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/subtractor_output_analyzer.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/suppression_filter.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/suppression_gain.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/suppression_gain_limiter.Plo \
./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/vector_buffer.Plo \
./webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/aecm_core.Plo \
./webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/aecm_core_c.Plo \
./webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/echo_control_mobile.Plo \
./webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/agc.Plo \
./webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/agc_manager_direct.Plo \
./webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/loudness_histogram.Plo \
./webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/utility.Plo \
./webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/analog_agc.Plo \
./webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/digital_agc.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/adaptive_agc.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/adaptive_digital_gain_applier.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/adaptive_mode_level_estimator.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/adaptive_mode_level_estimator_agc.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/agc2_common.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/agc2_testing_common.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/biquad_filter.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/compute_interpolated_gain_curve.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/down_sampler.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/fixed_digital_level_estimator.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/fixed_gain_controller.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/gain_applier.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/interpolated_gain_curve.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/limiter.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/limiter_db_gain_curve.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/noise_level_estimator.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/noise_spectrum_estimator.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/saturation_protector.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/signal_classifier.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/vad_with_level.Plo \
./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/vector_float_frame.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/features_extraction.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/fft_util.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/lp_residual.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/pitch_search.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/pitch_search_internal.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/rnn.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/spectral_features.Plo \
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/spectral_features_internal.Plo \
./webrtc_dsp/modules/audio_processing/audio_generator/$(DEPDIR)/file_audio_generator.Plo \
./webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/circular_buffer.Plo \
./webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/mean_variance_estimator.Plo \
./webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/moving_max.Plo \
./webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/normalized_covariance_estimator.Plo \
./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/aec_dump.Plo \
./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/audio_generator_factory.Plo \
./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/audio_processing.Plo \
./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/audio_processing_statistics.Plo \
./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/config.Plo \
./webrtc_dsp/modules/audio_processing/logging/$(DEPDIR)/apm_data_dumper.Plo \
./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/noise_suppression.Plo \
./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/noise_suppression_x.Plo \
./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/ns_core.Plo \
./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/nsx_core.Plo \
./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/nsx_core_c.Plo \
./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/moving_moments.Plo \
./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/transient_detector.Plo \
./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/transient_suppressor.Plo \
./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/wpd_node.Plo \
./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/wpd_tree.Plo \
./webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/block_mean_calculator.Plo \
./webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/delay_estimator.Plo \
./webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/delay_estimator_wrapper.Plo \
./webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/ooura_fft.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/gmm.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/pitch_based_vad.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/pitch_internal.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/pole_zero_filter.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/standalone_vad.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/vad_audio_proc.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/vad_circular_buffer.Plo \
./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/voice_activity_detector.Plo \
./webrtc_dsp/modules/third_party/fft/$(DEPDIR)/fft.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/checks.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/criticalsection.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/event.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/event_tracer.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/logging_webrtc.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/platform_file.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/platform_thread.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/platform_thread_types.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/race_checker.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/string_to_number.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/stringencode.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/stringutils.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/thread_checker_impl.Plo \
./webrtc_dsp/rtc_base/$(DEPDIR)/timeutils.Plo \
./webrtc_dsp/rtc_base/memory/$(DEPDIR)/aligned_malloc.Plo \
./webrtc_dsp/rtc_base/strings/$(DEPDIR)/string_builder.Plo \
./webrtc_dsp/system_wrappers/source/$(DEPDIR)/cpu_features.Plo \
./webrtc_dsp/system_wrappers/source/$(DEPDIR)/field_trial.Plo \
./webrtc_dsp/system_wrappers/source/$(DEPDIR)/metrics.Plo \
./webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/kiss_fft.Plo \
./webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/rnn_vad_weights.Plo \
audio/$(DEPDIR)/AudioIO.Plo \
audio/$(DEPDIR)/AudioIOCallback.Plo \
audio/$(DEPDIR)/AudioInput.Plo audio/$(DEPDIR)/AudioOutput.Plo \
audio/$(DEPDIR)/Resampler.Plo \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
os/darwin/$(DEPDIR)/AudioInputAudioUnit.Plo \
os/darwin/$(DEPDIR)/AudioInputAudioUnitOSX.Plo \
os/darwin/$(DEPDIR)/AudioOutputAudioUnit.Plo \
os/darwin/$(DEPDIR)/AudioOutputAudioUnitOSX.Plo \
os/darwin/$(DEPDIR)/AudioUnitIO.Plo \
os/darwin/$(DEPDIR)/DarwinSpecific.Plo \
os/linux/$(DEPDIR)/AudioInputALSA.Plo \
os/linux/$(DEPDIR)/AudioInputPulse.Plo \
os/linux/$(DEPDIR)/AudioOutputALSA.Plo \
os/linux/$(DEPDIR)/AudioOutputPulse.Plo \
os/linux/$(DEPDIR)/AudioPulse.Plo \
os/posix/$(DEPDIR)/NetworkSocketPosix.Plo \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
video/$(DEPDIR)/VideoRenderer.Plo \
video/$(DEPDIR)/VideoSource.Plo \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/complex_bit_reverse.Plo \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/complex_bit_reverse_arm.Plo \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/cross_correlation_neon.Plo \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/downsample_fast_neon.Plo \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/min_max_operations_neon.Plo \
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/spl_sqrt_floor.Plo \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/spl_sqrt_floor_arm.Plo \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/aec_core_neon.Plo \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/aec_core_sse2.Plo \
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/aecm_core_neon.Plo \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/nsx_core_neon.Plo \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/ooura_fft_neon.Plo \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/ooura_fft_sse2.Plo \
webrtc_dsp/rtc_base/$(DEPDIR)/logging_mac.Plo
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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am__nobase_tgvoipinclude_HEADERS_DIST = VoIPController.h Buffers.h \
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EchoCanceller.h JitterBuffer.h logging.h threading.h \
MediaStreamItf.h MessageThread.h NetworkSocket.h OpusDecoder.h \
OpusEncoder.h PacketReassembler.h VoIPServerConfig.h \
audio/AudioIO.h audio/AudioInput.h audio/AudioOutput.h \
audio/Resampler.h os/posix/NetworkSocketPosix.h \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
video/VideoSource.h video/VideoRenderer.h json11.hpp utils.h \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
os/darwin/AudioInputAudioUnit.h \
os/darwin/AudioOutputAudioUnit.h os/darwin/AudioUnitIO.h \
os/darwin/AudioInputAudioUnitOSX.h \
os/darwin/AudioOutputAudioUnitOSX.h os/darwin/DarwinSpecific.h \
os/linux/AudioInputALSA.h os/linux/AudioOutputALSA.h \
os/linux/AudioOutputPulse.h os/linux/AudioInputPulse.h \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/linux/AudioPulse.h os/linux/PulseFunctions.h \
audio/AudioIOCallback.h
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
CCASFLAGS = @CCASFLAGS@ $(am__append_9) $(am__append_20)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
CCDEPMODE = @CCDEPMODE@
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
$(am__append_11) $(am__append_13) $(am__append_15) \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
$(am__append_19) $(am__append_24)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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AUTOMAKE_OPTIONS = foreign
lib_LTLIBRARIES = libtgvoip.la
SRC = VoIPController.cpp Buffers.cpp CongestionControl.cpp \
EchoCanceller.cpp JitterBuffer.cpp logging.cpp \
MediaStreamItf.cpp MessageThread.cpp NetworkSocket.cpp \
OpusDecoder.cpp OpusEncoder.cpp PacketReassembler.cpp \
VoIPGroupController.cpp VoIPServerConfig.cpp audio/AudioIO.cpp \
audio/AudioInput.cpp audio/AudioOutput.cpp audio/Resampler.cpp \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/posix/NetworkSocketPosix.cpp video/VideoSource.cpp \
video/VideoRenderer.cpp json11.cpp $(am__append_1) \
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
$(am__append_4) $(am__append_6) $(am__append_10) \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
$(am__append_12) $(am__append_14) $(am__append_16) \
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
$(am__append_18) $(am__append_21) $(am__append_22) \
$(am__append_23)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
TGVOIP_HDRS = VoIPController.h Buffers.h BlockingQueue.h \
PrivateDefines.h CongestionControl.h EchoCanceller.h \
JitterBuffer.h logging.h threading.h MediaStreamItf.h \
MessageThread.h NetworkSocket.h OpusDecoder.h OpusEncoder.h \
PacketReassembler.h VoIPServerConfig.h audio/AudioIO.h \
audio/AudioInput.h audio/AudioOutput.h audio/Resampler.h \
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
os/posix/NetworkSocketPosix.h video/VideoSource.h \
video/VideoRenderer.h json11.hpp utils.h $(am__append_2) \
$(am__append_5) $(am__append_7) $(am__append_17)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
libtgvoip_la_SOURCES = $(SRC) $(TGVOIP_HDRS)
tgvoipincludedir = $(includedir)/tgvoip
nobase_tgvoipinclude_HEADERS = $(TGVOIP_HDRS)
all: config.h
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os/posix/NetworkSocketPosix.lo: os/posix/$(am__dirstamp) \
os/posix/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
video/$(am__dirstamp):
@$(MKDIR_P) video
@: > video/$(am__dirstamp)
video/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) video/$(DEPDIR)
@: > video/$(DEPDIR)/$(am__dirstamp)
video/VideoSource.lo: video/$(am__dirstamp) \
video/$(DEPDIR)/$(am__dirstamp)
video/VideoRenderer.lo: video/$(am__dirstamp) \
video/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
os/darwin/$(am__dirstamp):
@$(MKDIR_P) os/darwin
@: > os/darwin/$(am__dirstamp)
os/darwin/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) os/darwin/$(DEPDIR)
@: > os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/AudioInputAudioUnit.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/AudioOutputAudioUnit.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/AudioUnitIO.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/AudioInputAudioUnitOSX.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/AudioOutputAudioUnitOSX.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/darwin/DarwinSpecific.lo: os/darwin/$(am__dirstamp) \
os/darwin/$(DEPDIR)/$(am__dirstamp)
os/linux/$(am__dirstamp):
@$(MKDIR_P) os/linux
@: > os/linux/$(am__dirstamp)
os/linux/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) os/linux/$(DEPDIR)
@: > os/linux/$(DEPDIR)/$(am__dirstamp)
os/linux/AudioInputALSA.lo: os/linux/$(am__dirstamp) \
os/linux/$(DEPDIR)/$(am__dirstamp)
os/linux/AudioOutputALSA.lo: os/linux/$(am__dirstamp) \
os/linux/$(DEPDIR)/$(am__dirstamp)
os/linux/AudioOutputPulse.lo: os/linux/$(am__dirstamp) \
os/linux/$(DEPDIR)/$(am__dirstamp)
os/linux/AudioInputPulse.lo: os/linux/$(am__dirstamp) \
os/linux/$(DEPDIR)/$(am__dirstamp)
os/linux/AudioPulse.lo: os/linux/$(am__dirstamp) \
os/linux/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
webrtc_dsp/system_wrappers/source/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/system_wrappers/source
@: > webrtc_dsp/system_wrappers/source/$(am__dirstamp)
webrtc_dsp/system_wrappers/source/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/system_wrappers/source/$(DEPDIR)
@: > webrtc_dsp/system_wrappers/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/system_wrappers/source/field_trial.lo: \
webrtc_dsp/system_wrappers/source/$(am__dirstamp) \
webrtc_dsp/system_wrappers/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/system_wrappers/source/metrics.lo: \
webrtc_dsp/system_wrappers/source/$(am__dirstamp) \
webrtc_dsp/system_wrappers/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/system_wrappers/source/cpu_features.lo: \
webrtc_dsp/system_wrappers/source/$(am__dirstamp) \
webrtc_dsp/system_wrappers/source/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/absl/strings/internal/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/strings/internal
@: > webrtc_dsp/absl/strings/internal/$(am__dirstamp)
webrtc_dsp/absl/strings/internal/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/strings/internal/$(DEPDIR)
@: > webrtc_dsp/absl/strings/internal/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/strings/internal/memutil.lo: \
webrtc_dsp/absl/strings/internal/$(am__dirstamp) \
webrtc_dsp/absl/strings/internal/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/absl/strings/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/strings
@: > webrtc_dsp/absl/strings/$(am__dirstamp)
webrtc_dsp/absl/strings/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/strings/$(DEPDIR)
@: > webrtc_dsp/absl/strings/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/strings/string_view.lo: \
webrtc_dsp/absl/strings/$(am__dirstamp) \
webrtc_dsp/absl/strings/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/strings/ascii.lo: \
webrtc_dsp/absl/strings/$(am__dirstamp) \
webrtc_dsp/absl/strings/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/absl/types/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/types
@: > webrtc_dsp/absl/types/$(am__dirstamp)
webrtc_dsp/absl/types/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/types/$(DEPDIR)
@: > webrtc_dsp/absl/types/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/types/bad_optional_access.lo: \
webrtc_dsp/absl/types/$(am__dirstamp) \
webrtc_dsp/absl/types/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/types/optional.lo: \
webrtc_dsp/absl/types/$(am__dirstamp) \
webrtc_dsp/absl/types/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/absl/base/internal/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/base/internal
@: > webrtc_dsp/absl/base/internal/$(am__dirstamp)
webrtc_dsp/absl/base/internal/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/absl/base/internal/$(DEPDIR)
@: > webrtc_dsp/absl/base/internal/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/base/internal/raw_logging.lo: \
webrtc_dsp/absl/base/internal/$(am__dirstamp) \
webrtc_dsp/absl/base/internal/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/absl/base/internal/throw_delegate.lo: \
webrtc_dsp/absl/base/internal/$(am__dirstamp) \
webrtc_dsp/absl/base/internal/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/rtc_base/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base
@: > webrtc_dsp/rtc_base/$(am__dirstamp)
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base/$(DEPDIR)
@: > webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/race_checker.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/rtc_base/strings/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base/strings
@: > webrtc_dsp/rtc_base/strings/$(am__dirstamp)
webrtc_dsp/rtc_base/strings/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base/strings/$(DEPDIR)
@: > webrtc_dsp/rtc_base/strings/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/strings/string_builder.lo: \
webrtc_dsp/rtc_base/strings/$(am__dirstamp) \
webrtc_dsp/rtc_base/strings/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/rtc_base/memory/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base/memory
@: > webrtc_dsp/rtc_base/memory/$(am__dirstamp)
webrtc_dsp/rtc_base/memory/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/rtc_base/memory/$(DEPDIR)
@: > webrtc_dsp/rtc_base/memory/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/memory/aligned_malloc.lo: \
webrtc_dsp/rtc_base/memory/$(am__dirstamp) \
webrtc_dsp/rtc_base/memory/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/timeutils.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/platform_file.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/string_to_number.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/thread_checker_impl.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/stringencode.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/stringutils.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/checks.lo: webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/platform_thread.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/logging_webrtc.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/criticalsection.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/platform_thread_types.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/event.lo: webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/rtc_base/event_tracer.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/third_party/rnnoise/src/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/third_party/rnnoise/src
@: > webrtc_dsp/third_party/rnnoise/src/$(am__dirstamp)
webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)
@: > webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/third_party/rnnoise/src/rnn_vad_weights.lo: \
webrtc_dsp/third_party/rnnoise/src/$(am__dirstamp) \
webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/third_party/rnnoise/src/kiss_fft.lo: \
webrtc_dsp/third_party/rnnoise/src/$(am__dirstamp) \
webrtc_dsp/third_party/rnnoise/src/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/api/audio/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/api/audio
@: > webrtc_dsp/api/audio/$(am__dirstamp)
webrtc_dsp/api/audio/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/api/audio/$(DEPDIR)
@: > webrtc_dsp/api/audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/api/audio/audio_frame.lo: \
webrtc_dsp/api/audio/$(am__dirstamp) \
webrtc_dsp/api/audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/api/audio/echo_canceller3_config.lo: \
webrtc_dsp/api/audio/$(am__dirstamp) \
webrtc_dsp/api/audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/api/audio/echo_canceller3_factory.lo: \
webrtc_dsp/api/audio/$(am__dirstamp) \
webrtc_dsp/api/audio/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/third_party/fft/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/third_party/fft
@: > webrtc_dsp/modules/third_party/fft/$(am__dirstamp)
webrtc_dsp/modules/third_party/fft/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/third_party/fft/$(DEPDIR)
@: > webrtc_dsp/modules/third_party/fft/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/third_party/fft/fft.lo: \
webrtc_dsp/modules/third_party/fft/$(am__dirstamp) \
webrtc_dsp/modules/third_party/fft/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_coding/codecs/isac/main/source
@: > webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp)
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)
@: > webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_estimator.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filterbanks.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/transform.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_filter.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/filter_functions.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lattice.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/intialize.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/encode.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_analysis.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/entropy_coding.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac_vad.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/arith_routines.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/crc.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/decode_bwe.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_coding/codecs/isac/main/source/isac.lo: webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(am__dirstamp) \
webrtc_dsp/modules/audio_coding/codecs/isac/main/source/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing
@: > webrtc_dsp/modules/audio_processing/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/rms_level.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/echo_detector
@: > webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_detector/normalized_covariance_estimator.lo: webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_detector/moving_max.lo: webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_detector/circular_buffer.lo: webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_detector/mean_variance_estimator.lo: webrtc_dsp/modules/audio_processing/echo_detector/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/echo_detector/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/splitting_filter.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/gain_control_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/ns
@: > webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/ns/nsx_core.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/ns/noise_suppression_x.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/ns/nsx_core_c.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/ns/ns_core.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/ns/noise_suppression.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/audio_buffer.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/typing_detection.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/include
@: > webrtc_dsp/modules/audio_processing/include/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/include/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.lo: \
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/include/audio_generator_factory.lo: \
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/include/aec_dump.lo: \
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/include/audio_processing.lo: \
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/include/config.lo: \
webrtc_dsp/modules/audio_processing/include/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/include/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc2
@: > webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/interpolated_gain_curve.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/agc2_common.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/gain_applier.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/adaptive_agc.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/adaptive_digital_gain_applier.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/limiter.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/saturation_protector.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc2/rnn_vad
@: > webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/rnn.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/spectral_features.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/pitch_search.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/features_extraction.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/fft_util.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/rnn_vad/lp_residual.lo: webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/rnn_vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/vector_float_frame.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/noise_level_estimator.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/agc2_testing_common.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/fixed_digital_level_estimator.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/fixed_gain_controller.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/vad_with_level.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/limiter_db_gain_curve.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/down_sampler.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/signal_classifier.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/noise_spectrum_estimator.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/compute_interpolated_gain_curve.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/biquad_filter.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc2/adaptive_mode_level_estimator.lo: \
webrtc_dsp/modules/audio_processing/agc2/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc2/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/transient
@: > webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/transient/moving_moments.lo: \
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/transient/wpd_tree.lo: \
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/transient/wpd_node.lo: \
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/transient/transient_suppressor.lo: \
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/transient/transient_detector.lo: \
webrtc_dsp/modules/audio_processing/transient/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/transient/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/low_cut_filter.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/level_estimator_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/three_band_filter_bank.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aec
@: > webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec/echo_cancellation.lo: \
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec/aec_resampler.lo: \
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec/aec_core.lo: \
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/voice_detection_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_cancellation_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/gain_control_for_experimental_agc.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc
@: > webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/agc.lo: \
webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/loudness_histogram.lo: \
webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/agc_manager_direct.lo: \
webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc/legacy/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc/legacy
@: > webrtc_dsp/modules/audio_processing/agc/legacy/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/legacy/analog_agc.lo: webrtc_dsp/modules/audio_processing/agc/legacy/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/legacy/digital_agc.lo: webrtc_dsp/modules/audio_processing/agc/legacy/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/legacy/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/agc/utility.lo: \
webrtc_dsp/modules/audio_processing/agc/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/agc/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/audio_processing_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/audio_generator/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/audio_generator
@: > webrtc_dsp/modules/audio_processing/audio_generator/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/audio_generator/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/audio_generator/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/audio_generator/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/audio_generator/file_audio_generator.lo: webrtc_dsp/modules/audio_processing/audio_generator/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/audio_generator/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/gain_controller2.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/residual_echo_detector.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/noise_suppression_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aecm
@: > webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aecm/aecm_core.lo: \
webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aecm/aecm_core_c.lo: \
webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aecm/echo_control_mobile.lo: \
webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aec3
@: > webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_reverb_model.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_remover_metrics.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/matched_filter_lag_aggregator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer2.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_path_variability.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/frame_blocker.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/subtractor.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/aec3_fft.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/fullband_erle_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/suppression_filter.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/block_processor.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/subband_erle_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller_metrics.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_delay_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/vector_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/erl_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/aec_state.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/adaptive_fir_filter.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/skew_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_path_delay_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/block_framer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/erle_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/reverb_model.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/cascaded_biquad_filter.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/subtractor_output.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/stationarity_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_signal_analyzer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/subtractor_output_analyzer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/suppression_gain.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_audibility.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/block_processor_metrics.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/moving_average.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/reverb_model_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/aec3_common.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/residual_echo_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/matched_filter.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/reverb_decay_estimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/render_delay_controller2.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/main_filter_update_gain.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_remover.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/downsampled_render_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/matrix_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/block_processor2.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/echo_canceller3.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/block_delay_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/fft_buffer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/comfort_noise_generator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/shadow_filter_update_gain.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/filter_analyzer.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/reverb_frequency_response.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/aec3/decimator.lo: \
webrtc_dsp/modules/audio_processing/aec3/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec3/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/echo_control_mobile_impl.lo: \
webrtc_dsp/modules/audio_processing/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/logging/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/logging
@: > webrtc_dsp/modules/audio_processing/logging/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/logging/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/logging/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/logging/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/logging/apm_data_dumper.lo: \
webrtc_dsp/modules/audio_processing/logging/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/logging/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/vad
@: > webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/voice_activity_detector.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/standalone_vad.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/pitch_internal.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/vad_circular_buffer.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/vad_audio_proc.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/pole_zero_filter.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/pitch_based_vad.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/vad/gmm.lo: \
webrtc_dsp/modules/audio_processing/vad/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/vad/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/utility
@: > webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)
@: > webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/utility/ooura_fft.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/utility/delay_estimator_wrapper.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/utility/delay_estimator.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/modules/audio_processing/utility/block_mean_calculator.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio
@: > webrtc_dsp/common_audio/$(am__dirstamp)
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/$(DEPDIR)
@: > webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/window_generator.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/channel_buffer.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/fir_filter_factory.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/wav_header.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/real_fourier_ooura.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/audio_util.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/fir_filter_sse.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/resampler/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/resampler
@: > webrtc_dsp/common_audio/resampler/$(am__dirstamp)
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/resampler/$(DEPDIR)
@: > webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/push_sinc_resampler.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/resampler.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/sinc_resampler_sse.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/push_resampler.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/sinc_resampler.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/resampler/sinusoidal_linear_chirp_source.lo: \
webrtc_dsp/common_audio/resampler/$(am__dirstamp) \
webrtc_dsp/common_audio/resampler/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/wav_file.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/third_party/fft4g/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/third_party/fft4g
@: > webrtc_dsp/common_audio/third_party/fft4g/$(am__dirstamp)
webrtc_dsp/common_audio/third_party/fft4g/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/third_party/fft4g/$(DEPDIR)
@: > webrtc_dsp/common_audio/third_party/fft4g/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/third_party/fft4g/fft4g.lo: \
webrtc_dsp/common_audio/third_party/fft4g/$(am__dirstamp) \
webrtc_dsp/common_audio/third_party/fft4g/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/audio_converter.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/real_fourier.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/sparse_fir_filter.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/smoothing_filter.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/fir_filter_c.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/ring_buffer.lo: \
webrtc_dsp/common_audio/$(am__dirstamp) \
webrtc_dsp/common_audio/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/signal_processing
@: > webrtc_dsp/common_audio/signal_processing/$(am__dirstamp)
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/signal_processing/$(DEPDIR)
@: > webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/complex_fft.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/filter_ma_fast_q12.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/splitting_filter1.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/levinson_durbin.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/dot_product_with_scale.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/auto_corr_to_refl_coef.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/resample_by_2_internal.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/energy.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/sqrt_of_one_minus_x_squared.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/downsample_fast.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/spl_init.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/lpc_to_refl_coef.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/cross_correlation.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/division_operations.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/auto_correlation.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/get_scaling_square.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/resample.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/min_max_operations.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/refl_coef_to_lpc.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/filter_ar.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/vector_scaling_operations.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/resample_fractional.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/real_fft.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/ilbc_specific_functions.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/randomization_functions.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/copy_set_operations.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/resample_by_2.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/get_hanning_window.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/resample_48khz.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/spl_inl.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/signal_processing/spl_sqrt.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/vad/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/vad
@: > webrtc_dsp/common_audio/vad/$(am__dirstamp)
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) ./webrtc_dsp/common_audio/vad/$(DEPDIR)
@: > webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/vad_sp.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/vad.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/webrtc_vad.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/vad_filterbank.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/vad_core.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
./webrtc_dsp/common_audio/vad/vad_gmm.lo: \
webrtc_dsp/common_audio/vad/$(am__dirstamp) \
webrtc_dsp/common_audio/vad/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/rtc_base/logging_mac.lo: \
webrtc_dsp/rtc_base/$(am__dirstamp) \
webrtc_dsp/rtc_base/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec/aec_core_sse2.lo: \
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
audio/AudioIOCallback.lo: audio/$(am__dirstamp) \
audio/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/signal_processing/complex_bit_reverse_arm.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(am__dirstamp):
@$(MKDIR_P) webrtc_dsp/common_audio/third_party/spl_sqrt_floor
@: > webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(am__dirstamp)
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/$(am__dirstamp):
@$(MKDIR_P) webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)
@: > webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.lo: webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(am__dirstamp) \
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
webrtc_dsp/common_audio/signal_processing/cross_correlation_neon.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/signal_processing/downsample_fast_neon.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/signal_processing/min_max_operations_neon.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aec/aec_core_neon.lo: \
webrtc_dsp/modules/audio_processing/aec/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/aecm/aecm_core_neon.lo: \
webrtc_dsp/modules/audio_processing/aecm/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/aecm/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/ns/nsx_core_neon.lo: \
webrtc_dsp/modules/audio_processing/ns/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/ns/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/modules/audio_processing/utility/ooura_fft_neon.lo: \
webrtc_dsp/modules/audio_processing/utility/$(am__dirstamp) \
webrtc_dsp/modules/audio_processing/utility/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
2019-02-14 07:27:45 +00:00
webrtc_dsp/common_audio/signal_processing/complex_bit_reverse.lo: \
webrtc_dsp/common_audio/signal_processing/$(am__dirstamp) \
webrtc_dsp/common_audio/signal_processing/$(DEPDIR)/$(am__dirstamp)
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.lo: webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(am__dirstamp) \
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/$(DEPDIR)/$(am__dirstamp)
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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mostlyclean-compile:
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
distclean-compile:
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
-rm -rf video/.libs video/_libs
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-rm -rf webrtc_dsp/common_audio/third_party/spl_sqrt_floor/.libs webrtc_dsp/common_audio/third_party/spl_sqrt_floor/_libs
-rm -rf webrtc_dsp/modules/audio_processing/aec/.libs webrtc_dsp/modules/audio_processing/aec/_libs
-rm -rf webrtc_dsp/modules/audio_processing/aecm/.libs webrtc_dsp/modules/audio_processing/aecm/_libs
-rm -rf webrtc_dsp/modules/audio_processing/ns/.libs webrtc_dsp/modules/audio_processing/ns/_libs
-rm -rf webrtc_dsp/modules/audio_processing/utility/.libs webrtc_dsp/modules/audio_processing/utility/_libs
-rm -rf webrtc_dsp/rtc_base/.libs webrtc_dsp/rtc_base/_libs
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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maintainer-clean-generic:
@echo "This command is intended for maintainers to use"
@echo "it deletes files that may require special tools to rebuild."
clean: clean-am
clean-am: clean-generic clean-libLTLIBRARIES clean-libtool \
mostlyclean-am
distclean: distclean-am
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
-rm -f audio/$(DEPDIR)/AudioIO.Plo
Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
-rm -f video/$(DEPDIR)/VideoRenderer.Plo
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Squashed 'libtgvoip/' changes from fc13464..6e7384d 6e7384d Debug stats a58c5fb Fixed build with TGVOIP_NO_DSP 1b699b3 p2p improvements be5f739 Merge pull request #73 from bakatrouble/public 5efda7a Fix ARM Linux builds git-subtree-dir: libtgvoip git-subtree-split: 6e7384d5a8ddf01cc159392a69a49d816434282c
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
-rm -f webrtc_dsp/modules/audio_processing/aec/$(DEPDIR)/aec_core_neon.Plo
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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-rm -f Makefile
distclean-am: clean-am distclean-compile distclean-generic \
distclean-hdr distclean-libtool distclean-tags
dvi: dvi-am
dvi-am:
html: html-am
html-am:
info: info-am
info-am:
install-data-am: install-nobase_tgvoipincludeHEADERS
install-dvi: install-dvi-am
install-dvi-am:
install-exec-am: install-libLTLIBRARIES
install-html: install-html-am
install-html-am:
install-info: install-info-am
install-info-am:
install-man:
install-pdf: install-pdf-am
install-pdf-am:
install-ps: install-ps-am
install-ps-am:
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
2019-02-06 18:22:38 +00:00
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Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
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Squashed 'libtgvoip/' changes from 6053cf5..cfd62e6 cfd62e6 Why did it change the OS X project 3a58a16 2.4.3 c4a48b3 Updated OS X project 564eada Fix #63 4f64e2e fixes 0c732e2 fixes 12e76ed better logging f015b79 Merge pull request #62 from xvitaly/big-endian a1df90f Set preferred audio session parameters on iOS 59a975b Fixes 8fd89fc Fixes, mic level testing and volume adjustment 243acfa Backported WebRTC upstream patch with Big Endian support. fed3bb7 Detect when proxy does not support UDP and persist that across calls a7546d4 Merge commit '6d03dd9ae4bf48d7344341cdd2d055ebd3a6a42e' into public 6d03dd9 version 69adf70 Use server config for APM + iOS crash fix 0b42ec8 Update iOS project f1b9e63 packet logging beeea45 I apparently still suck at C++ memory management 24fceba Update project 7f54b91 crash fix f85ce99 Save more data in data saving mode f4c4f79 Collect packet stats and accept json string for server config 78e584c New protocol version: optimized packet size 8cf9177 Fixed build on iOS 9dd089d fixed build on android 5caaaaf Updated WebRTC APM cc0cf35 fixed deadlock 02f4835 Rearranged VoIPController methods and added sections 912f73d Updated OS X project 39376df Fixed audio glitches on Windows dfe1f03 Updated project 81daf3f fix 296187a Merge pull request #58 from telegramdesktop/tdesktop 44956ac Merge pull request #57 from UnigramDev/public fb0a2b0 Fix build for Linux. d6cf1b7 Updated UWP wrapper 0f06289 Merge branch 'public' of github.com:grishka/libtgvoip into public dcfad91 Fix #54 162f447 Merge pull request #56 from telegramdesktop/tdesktop a7ee511 Merge remote-tracking branch 'origin/tdesktop' into HEAD 467b148 Removed unused files b1a0b3d 2.3 9b292fd Fix warning in Xcode 10. 8d8522a Merge pull request #53 from UnigramDev/public 646f7d6 Merge branch 'public' into public 14d782b Fixes 68acf59 Added GetSignalBarsCount and GetConnectionState to CXWrapper 761c586 Added GetStats to CXWrapper f643b02 Prevent crash if UWP WASAPI devices aren't found b2ac10e Fixed UWP project 9a1ec51 Fixed build for Windows Phone, fixed some warnings 4aea54f fix git-subtree-dir: libtgvoip git-subtree-split: cfd62e66a825348ac51f49e5d20bf8827fef7a38
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mostlyclean: mostlyclean-am
mostlyclean-am: mostlyclean-compile mostlyclean-generic \
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pdf: pdf-am
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.PRECIOUS: Makefile
# Tell versions [3.59,3.63) of GNU make to not export all variables.
# Otherwise a system limit (for SysV at least) may be exceeded.
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