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webrtc/modules/audio_processing
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Gustaf Ullberg ddb82a6b5f AEC3: Fix filter output transition when input and output is the same array 
This CL fixes a bug in the filter output transition when the 'from' input
points to the same array as the output. It also includes a slight
improvement to the transition by starting one sample earlier than
previously.

Bug: webrtc:9741,chromium:882789
Change-Id: Ifd5f16c1ac88a74d93499e7f4b4c0e5cb3e4976f
Reviewed-on: https://webrtc-review.googlesource.com/99540
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24683}
2018-09-11 11:59:12 +00:00
..
aec Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
aec3 AEC3: Fix filter output transition when input and output is the same array 2018-09-11 11:59:12 +00:00
aec_dump Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
aecm Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
agc Gain metrics for digital adaptive AGC. 2018-08-15 13:44:46 +00:00
agc2 Fix no_global_constructors in audio_processing/agc2/rnn_vad. 2018-09-11 06:23:56 +00:00
audio_generator Add stub draft of audio generator to APM 2018-03-05 09:28:52 +00:00
echo_detector Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
include Added flags for the adaptive analog AGC in audioproc_f. 2018-09-10 14:16:46 +00:00
logging Remove stringstream usages from the APM 2018-04-06 14:17:03 +00:00
ns Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
test Added flags for the adaptive analog AGC in audioproc_f. 2018-09-10 14:16:46 +00:00
transient Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
utility Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
vad Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
audio_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_buffer.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_frame_view_unittest.cc Add namespace 'webrtc' to AudioFrameView. 2018-05-14 12:33:49 +00:00
audio_processing_impl.cc Added flags for the adaptive analog AGC in audioproc_f. 2018-09-10 14:16:46 +00:00
audio_processing_impl.h Added flags for the adaptive analog AGC in audioproc_f. 2018-09-10 14:16:46 +00:00
audio_processing_impl_locking_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing_impl_unittest.cc APM: render pre-processor moved before echo detector queuing. 2018-08-09 14:40:31 +00:00
audio_processing_performance_unittest.cc Removing the intelligibility enhancer. 2018-08-30 21:29:57 +00:00
audio_processing_unittest.cc Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
BUILD.gn Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +00:00
common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
debug.proto Store RuntimeSetting in Aec Dumps. 2018-09-10 11:40:28 +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.cc Set AEC2 suppression to high by default 2018-08-16 12:14:14 +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_cancellation_impl_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
echo_cancellation_proxy.cc Add AEC proxies for simple deprecation of AEC configurability. 2018-08-16 15:16:44 +00:00
echo_cancellation_proxy.h Add AEC proxies for simple deprecation of AEC configurability. 2018-08-16 15:16:44 +00:00
echo_control_mobile_impl.cc Turn off comfort noise generation by default in AECM 2018-07-24 08:52:36 +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_proxy.cc Add AEC proxies for simple deprecation of AEC configurability. 2018-08-16 15:16:44 +00:00
echo_control_mobile_proxy.h Add AEC proxies for simple deprecation of AEC configurability. 2018-08-16 15:16:44 +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 Atomically increment GainControl instance counter. 2018-08-15 07:44:00 +00:00
gain_control_for_experimental_agc.h Reset Agc2 on analog gain changes. 2018-08-08 14:36:37 +00:00
gain_control_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
gain_control_impl.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +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.cc Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
gain_controller2.h Optionally disable digital adaptive AGC2. 2018-08-24 15:54:43 +00:00
gain_controller2_unittest.cc Set a positive initial gain in the Adaptive Digital GC. 2018-04-27 09:05:25 +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.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
low_cut_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Adding alessiob@ and minyue@ as owners of APM. 2018-07-02 07:45:31 +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.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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
residual_echo_detector_unittest.cc Change echo detector to scoped_refptr 2018-06-14 09:51:41 +00:00
rms_level.cc Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
rms_level.h Delete root header file typedef.h. 2018-07-25 14:59:26 +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
splitting_filter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
splitting_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
typing_detection.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
voice_detection_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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