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webrtc/modules/audio_processing/test
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Gustaf Ullberg 41dd22b15d AEC3: Removing more dead code from the suppressor 
This CL removes the UpdateGainIncrease code that is not used anymore.
The CL has been tested for bit exactness.

Bug: webrtc:8671
Change-Id: I4fcf26c3b4b5bba760ee139416ddefac86a36c2e
Reviewed-on: https://webrtc-review.googlesource.com/95940
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24425}
2018-08-24 10:25:00 +00:00
..
android/apmtest Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
conversational_speech Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
py_quality_assessment Fixing py lint errors 2018-07-23 15:28:48 +00:00
aec_dump_based_simulator.cc Correct audioproc_f to support the new echo canceller activation III 2018-08-23 13:48:33 +00:00
aec_dump_based_simulator.h Restructure the audioproc_f tool into a library with a thin executable wrapper. 2018-03-09 18:06:04 +00:00
apmtest.m Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
audio_buffer_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer_tools.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_processing_simulator.cc AEC3: Removing more dead code from the suppressor 2018-08-24 10:25:00 +00:00
audio_processing_simulator.h Adding quiet mode for audioproc_f 2018-08-24 05:52:43 +00:00
audioproc_float_impl.cc Adding quiet mode for audioproc_f 2018-08-24 05:52:43 +00:00
audioproc_float_impl.h Moved audioproc_f interface into api directory. 2018-03-15 12:31:37 +00:00
audioproc_float_main.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
bitexactness_tools.cc Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. 2018-01-12 15:27:50 +00:00
bitexactness_tools.h Add DCHECK and documentation to disallow trying to read more than two audio channels in helper function. 2018-01-12 15:27:50 +00:00
debug_dump_replayer.cc Toggle AECs via AudioProcessing::Config 2018-08-17 14:56:57 +00:00
debug_dump_replayer.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
debug_dump_test.cc Toggle AECs via AudioProcessing::Config 2018-08-17 14:56:57 +00:00
echo_canceller_test_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_canceller_test_tools.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_canceller_test_tools_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fake_recording_device.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
fake_recording_device.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fake_recording_device_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
performance_timer.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
performance_timer.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
protobuf_utils.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
protobuf_utils.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulator_buffers.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
simulator_buffers.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
test_utils.cc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
test_utils.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
unittest.proto Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
wav_based_simulator.cc Added an audioproc option to not report the stream delay 2018-05-28 13:22:29 +00:00
wav_based_simulator.h Restructure the audioproc_f tool into a library with a thin executable wrapper. 2018-03-09 18:06:04 +00:00