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webrtc_m130/modules/audio_processing
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Alex Loiko 623472219f Store RuntimeSetting in Aec Dumps. 
Also read and apply settings when parsing and replaying dumps.

The implementation contains
* an extra field in debug.proto for the runtime settings
* code in AudioProcessingImpl to initiate the logging of the RS to the
  AecDump
* code in aec_dump/ to log the RS in the AecDump
* code in test/ for re-playing the RS. E.g. for APM simulation with
  audioproc_f.

Bug: webrtc:9138
Change-Id: Ia2a00537c2eb19484ff442fbffd0b95f8495516f
Reviewed-on: https://webrtc-review.googlesource.com/70502
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24647}
2018-09-10 11:40:28 +00:00
..
aec
Explicitly add -mfpu=neon to all targets that use NEON
2018-08-01 13:15:42 +00:00
aec3
AEC3: Parametrize the delay estimator to leverage strong echo paths
2018-09-06 23:01:58 +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
Revert "Fix no_global_constructors in audio_processing/agc2/rnn_vad."
2018-09-07 13:34:39 +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
Store RuntimeSetting in Aec Dumps.
2018-09-10 11:40:28 +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
Store RuntimeSetting in Aec Dumps.
2018-09-10 11:40:28 +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_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_frame_view_unittest.cc
Add namespace 'webrtc' to AudioFrameView.
2018-05-14 12:33:49 +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_impl.cc
Store RuntimeSetting in Aec Dumps.
2018-09-10 11:40:28 +00:00
audio_processing_impl.h
Removing the intelligibility enhancer.
2018-08-30 21:29:57 +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_unittest.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +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_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_unittest.cc
Set a positive initial gain in the Adaptive Digital GC.
2018-04-27 09:05:25 +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
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
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
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_unittest.cc
Change echo detector to scoped_refptr
2018-06-14 09:51:41 +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
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
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
splitting_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +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
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