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libtgvoip/Makefile.am

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2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
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 \
Collect packet stats and accept json string for server config
2018-11-29 01:03:15 +01:00
os/posix/NetworkSocketPosix.cpp \
video/VideoSource.cpp \
video/VideoRenderer.cpp \
Updated projects
2019-03-12 14:15:22 +01:00
video/ScreamCongestionController.cpp \
Collect packet stats and accept json string for server config
2018-11-29 01:03:15 +01:00
json11.cpp
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
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 \
Collect packet stats and accept json string for server config
2018-11-29 01:03:15 +01:00
os/posix/NetworkSocketPosix.h \
video/VideoSource.h \
video/VideoRenderer.h \
Updated projects
2019-03-12 14:15:22 +01:00
video/ScreamCongestionController.h \
Fix #63
2019-01-11 00:25:04 +01:00
json11.hpp \
utils.h
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
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 \
Updated projects
2019-03-12 14:15:22 +01:00
os/darwin/DarwinSpecific.mm \
os/darwin/SampleBufferDisplayLayerRenderer.mm \
os/darwin/TGVVideoRenderer.mm \
os/darwin/TGVVideoSource.mm \
os/darwin/VideoToolboxEncoderSource.mm
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
TGVOIP_HDRS += \
os/darwin/AudioInputAudioUnit.h \
os/darwin/AudioOutputAudioUnit.h \
os/darwin/AudioUnitIO.h \
os/darwin/AudioInputAudioUnitOSX.h \
os/darwin/AudioOutputAudioUnitOSX.h \
Updated projects
2019-03-12 14:15:22 +01:00
os/darwin/DarwinSpecific.h \
os/darwin/SampleBufferDisplayLayerRenderer.h \
os/darwin/TGVVideoRenderer.h \
os/darwin/TGVVideoSource.h \
os/darwin/VideoToolboxEncoderSource.h
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
Updated projects
2019-03-12 14:15:22 +01:00
LDFLAGS += -framework Foundation -framework CoreFoundation -framework CoreAudio -framework AudioToolbox -framework VideoToolbox -framework CoreMedia -framework CoreVideo
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
else
# Linux-specific
Updated projects
2019-03-12 14:15:22 +01:00
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
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
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
CFLAGS += -DWEBRTC_POSIX -DWEBRTC_APM_DEBUG_DUMP=0 -DWEBRTC_NS_FLOAT -I$(top_srcdir)/webrtc_dsp
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
CCASFLAGS += -I$(top_srcdir)/webrtc_dsp
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
./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 \
Fix ARM Linux builds
2019-02-10 12:04:22 +01:00
./webrtc_dsp/common_audio/vad/vad_gmm.c
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
if TARGET_OS_OSX
CFLAGS += -DWEBRTC_MAC
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
webrtc_dsp/rtc_base/logging_mac.mm \
webrtc_dsp/rtc_base/logging_mac.h
else
CFLAGS += -DWEBRTC_LINUX
endif
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
if TARGET_CPU_X86
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
webrtc_dsp/modules/audio_processing/aec/aec_core_sse2.cc \
webrtc_dsp/modules/audio_processing/utility/ooura_fft_sse2.cc
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
endif
added --enable-audio-callback to configure
2018-09-02 14:43:49 +02:00
if ENABLE_AUDIO_CALLBACK
CFLAGS += -DTGVOIP_USE_CALLBACK_AUDIO_IO
SRC += \
audio/AudioIOCallback.cpp
TGVOIP_HDRS += \
audio/AudioIOCallback.h
endif
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
if TARGET_CPU_ARM
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
webrtc_dsp/common_audio/signal_processing/complex_bit_reverse_arm.S \
Fix ARM Linux builds
2019-02-10 12:04:22 +01:00
webrtc_dsp/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
if TARGET_CPU_ARMV7
CFLAGS += -mfpu=neon -mfloat-abi=hard
CCASFLAGS += -mfpu=neon -mfloat-abi=hard
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
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
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
endif
Fix ARM Linux builds
2019-02-10 12:04:22 +01:00
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
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
endif
# headers
SRC += \
Updated WebRTC APM I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
2018-11-21 19:22:31 +01:00
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
2.2 - Refactored audio I/O to allow sharing a common context between input and output, for those OSes that require this - Rewritten periodic operation handling to use a "run loop" thingy instead of an ugly loop formerly known as tick thread - Fixed a bunch of compiler warnings (closes #13) - Added automake so you no longer need to use the GYP file for standalone builds (closes #43)
2018-07-17 18:48:21 +02:00
else
CFLAGS += -DTGVOIP_NO_DSP
endif
libtgvoip_la_SOURCES = $(SRC) $(TGVOIP_HDRS)
tgvoipincludedir = $(includedir)/tgvoip
nobase_tgvoipinclude_HEADERS = $(TGVOIP_HDRS)
Updated projects
2019-03-12 14:15:22 +01:00
CXXFLAGS += -std=gnu++0x $(CFLAGS)
if TARGET_OS_OSX
OBJCFLAGS = $(CFLAGS)
OBJCXXFLAGS += -std=gnu++0x $(CFLAGS)
endif
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