webrtc/modules/audio_processing
Gustaf Ullberg c4b7f037b7 AEC3: Adjust active render limits for downsampling factor 8
The signal used for delay estimation at downsampling factor 8 is bandpass
filtered and contains less energy than for other downsampling factors.
This CL adjusts the energy threshold used for determining if there is enough
farend activity to update the matched filters in the delay estimator.
Only downsampling factor 8 is affected.

Bug: webrtc:9288,chromium:846615
Change-Id: I6f38f5609a31e7a08e60571ac75ea75c9962e026
Reviewed-on: https://webrtc-review.googlesource.com/80443
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23486}
2018-06-01 10:07:16 +00:00
..
aec Delete unneeded includes of wav_file.h and file_wrapper.h. 2018-03-20 15:59:27 +00:00
aec3 AEC3: Adjust active render limits for downsampling factor 8 2018-06-01 10:07:16 +00:00
aec_dump Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
aecm Use of unititialized value in AECM. 2018-01-25 15:09:14 +00:00
agc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
agc2 Delete unneeded includes of basictypes.h. 2018-05-21 19:35:08 +00:00
audio_generator Add stub draft of audio generator to APM 2018-03-05 09:28:52 +00:00
beamformer Fix macro clash with _USE_MATH_DEFINES. 2017-12-13 09:39:20 +00:00
echo_detector Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
include Delete macro RTC_DEFINE_STATIC_LOCAL. 2018-05-24 08:10:35 +00:00
intelligibility Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
logging Remove stringstream usages from the APM 2018-04-06 14:17:03 +00:00
ns Fixing -Wstrict-prototypes warnings. 2018-03-19 16:57:21 +00:00
test AEC3: Added a mode to properly utilize highly linear setups 2018-05-29 07:59:03 +00:00
transient Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
utility Fix typo in the include path of ooura_fft.h 2018-01-11 07:57:40 +00:00
vad Break out the part of the iSAC codec that's used for Voice Activity Detection 2018-05-04 08:53:34 +00:00
audio_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_frame_view_unittest.cc Add namespace 'webrtc' to AudioFrameView. 2018-05-14 12:33:49 +00:00
audio_processing_impl_locking_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_processing_impl_unittest.cc Activate the pre-amplifier in AudioProcessing. 2018-04-16 14:36:49 +00:00
audio_processing_impl.cc Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
audio_processing_impl.h Add RuntimeSettings to CustomProcessing. 2018-05-15 10:03:25 +00:00
audio_processing_performance_unittest.cc Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_processing_unittest.cc Add RuntimeSettings to CustomProcessing. 2018-05-15 10:03:25 +00:00
BUILD.gn Merge :audio_processing and :aec_dump_interface. 2018-05-15 14:22:52 +00:00
common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug.proto Options and settings for the Pre-amplifier. 2018-04-16 12:25:48 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_bit_exact_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_impl_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
echo_cancellation_impl.cc Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS 2017-12-22 15:42:13 +00:00
echo_cancellation_impl.h Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS 2017-12-22 15:42:13 +00:00
echo_control_mobile_impl.cc Streamline error handling and logging in the audio processing module 2018-02-15 15:06:26 +00:00
echo_control_mobile_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_control_mobile_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_impl.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
gain_control_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_controller2_unittest.cc Set a positive initial gain in the Adaptive Digital GC. 2018-04-27 09:05:25 +00:00
gain_controller2.cc Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +00:00
gain_controller2.h Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +00:00
level_estimator_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add gustaf to audio_processing OWNERS 2018-02-06 10:54:29 +00:00
render_queue_item_verifier.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
residual_echo_detector_unittest.cc Make the echo detector injectable. 2018-01-11 15:43:01 +00:00
residual_echo_detector.cc Add more parameters to the Initialize function of the echo detector. 2018-03-15 09:21:56 +00:00
residual_echo_detector.h Add more parameters to the Initialize function of the echo detector. 2018-03-15 09:21:56 +00:00
rms_level_unittest.cc Move some more numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 12:39:39 +00:00
rms_level.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
rms_level.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
splitting_filter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
three_band_filter_bank.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
three_band_filter_bank.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
typing_detection.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
typing_detection.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
voice_detection_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
voice_detection_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
voice_detection_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00