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tg2sip/libtgvoip/Makefile.am

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Makefile
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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
AUTOMAKE_OPTIONS = foreign
CFLAGS = -Wall -DHAVE_CONFIG_H -Wno-unknown-pragmas
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
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
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
if TARGET_OS_OSX
SRC += \
os/darwin/AudioInputAudioUnit.cpp \
os/darwin/AudioOutputAudioUnit.cpp \
os/darwin/AudioUnitIO.cpp \
os/darwin/AudioInputAudioUnitOSX.cpp \
os/darwin/AudioOutputAudioUnitOSX.cpp \
os/darwin/DarwinSpecific.mm
TGVOIP_HDRS += \
os/darwin/AudioInputAudioUnit.h \
os/darwin/AudioOutputAudioUnit.h \
os/darwin/AudioUnitIO.h \
os/darwin/AudioInputAudioUnitOSX.h \
os/darwin/AudioOutputAudioUnitOSX.h \
os/darwin/DarwinSpecific.h
LDFLAGS += -framework Foundation -framework CoreFoundation -framework CoreAudio -framework AudioToolbox
else
# Linux-specific
if WITH_ALSA
SRC += \
os/linux/AudioInputALSA.cpp \
os/linux/AudioOutputALSA.cpp
TGVOIP_HDRS += \
os/linux/AudioInputALSA.h \
os/linux/AudioOutputALSA.h
endif
if WITH_PULSE
SRC += \
os/linux/AudioOutputPulse.cpp \
os/linux/AudioInputPulse.cpp \
os/linux/AudioPulse.cpp
TGVOIP_HDRS += \
os/linux/AudioOutputPulse.h \
os/linux/AudioInputPulse.h \
os/linux/AudioPulse.h \
os/linux/PulseFunctions.h
endif
endif
if ENABLE_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
CFLAGS += -DWEBRTC_POSIX -DWEBRTC_APM_DEBUG_DUMP=0 -DWEBRTC_NS_FLOAT -I$(top_srcdir)/webrtc_dsp
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
CCASFLAGS += -I$(top_srcdir)/webrtc_dsp
SRC += \
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 \
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/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
if TARGET_OS_OSX
CFLAGS += -DWEBRTC_MAC
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
SRC += \
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/rtc_base/logging_mac.mm \
webrtc_dsp/rtc_base/logging_mac.h
else
CFLAGS += -DWEBRTC_LINUX
endif
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
if TARGET_CPU_X86
SRC += \
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/aec_core_sse2.cc \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.cc
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
endif
if ENABLE_AUDIO_CALLBACK
CFLAGS += -DTGVOIP_USE_CALLBACK_AUDIO_IO
SRC += \
audio/AudioIOCallback.cpp
TGVOIP_HDRS += \
audio/AudioIOCallback.h
endif
if TARGET_CPU_ARM
SRC += \
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/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/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
if TARGET_CPU_ARMV7
CFLAGS += -mfpu=neon -mfloat-abi=hard
CCASFLAGS += -mfpu=neon -mfloat-abi=hard
SRC += \
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
# webrtc_dsp/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
endif
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
else
SRC += \
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/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
endif
# headers
SRC += \
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
Squashed 'libtgvoip/' content from commit 6053cf5 git-subtree-dir: libtgvoip git-subtree-split: 6053cf5dce58c12ef6ce2d2f6cbe77fd5b10c575
2018-10-03 14:34:43 +00:00
else
CFLAGS += -DTGVOIP_NO_DSP
endif
libtgvoip_la_SOURCES = $(SRC) $(TGVOIP_HDRS)
tgvoipincludedir = $(includedir)/tgvoip
nobase_tgvoipinclude_HEADERS = $(TGVOIP_HDRS)
CXXFLAGS += -std=gnu++0x $(CFLAGS)