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webrtc/modules/audio_processing/test
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Per Åhgren aa91b3c67e Hooks up more AEC3 parameters to be read by the AEC3 configuration file 
Bug: webrtc:8671
Change-Id: I593ea4965ab2f8215e5d55e0778caf83cf62d4e1
Reviewed-on: https://webrtc-review.googlesource.com/94480
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24304}
2018-08-16 08:30:48 +00:00
..
android/apmtest
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 Added an audioproc option to not report the stream delay 2018-05-28 13:22:29 +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
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 Hooks up more AEC3 parameters to be read by the AEC3 configuration file 2018-08-16 08:30:48 +00:00
audio_processing_simulator.h Optionally disable digital gain control in ExperimentalAgc. 2018-08-09 13:37:30 +00:00
audioproc_float_impl.cc Optionally disable digital gain control in ExperimentalAgc. 2018-08-09 13:37:30 +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 Revert "Add one-stop-shop for built-in AEC toggling in APM" 2018-07-23 14:48:17 +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 Revert "Add one-stop-shop for built-in AEC toggling in APM" 2018-07-23 14:48:17 +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
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