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webrtc/modules/audio_processing/test
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Alessio Bazzica 45adbafefe APM-QA unit test bug fix 
- temporary wav files created in temporary folder in TestExport.setUp()
- rename TestEchoPathSimulators -> TestExport

TBR=

Bug: webrtc:7494
Change-Id: I5b0c0675f539888e7392728055842c7772185921
Reviewed-on: https://webrtc-review.googlesource.com/14842
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20406}
2017-10-24 10:23:08 +00:00
..
android/apmtest
conversational_speech Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
py_quality_assessment APM-QA unit test bug fix 2017-10-24 10:23:08 +00:00
aec_dump_based_simulator.cc Separate AEC3 config from AudioProcessing::Config. 2017-10-19 08:19:52 +00:00
aec_dump_based_simulator.h audioproc_f with simulated mic analog gain 2017-09-27 10:27:56 +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 Separate AEC3 config from AudioProcessing::Config. 2017-10-19 08:19:52 +00:00
audio_processing_simulator.h AGC2 dummy module: fixed gain param, APM integration, audioproc_f adaptation 2017-10-13 11:05:37 +00:00
audioproc_float.cc AGC2 dummy module: fixed gain param, APM integration, audioproc_f adaptation 2017-10-13 11:05:37 +00:00
bitexactness_tools.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
bitexactness_tools.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug_dump_replayer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug_dump_replayer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug_dump_test.cc Separate AEC3 config from AudioProcessing::Config. 2017-10-19 08:19:52 +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 audioproc_f with simulated mic analog gain 2017-09-27 10:27:56 +00:00
fake_recording_device.h audioproc_f with simulated mic analog gain 2017-09-27 10:27:56 +00:00
fake_recording_device_unittest.cc audioproc_f with simulated mic analog gain 2017-09-27 10:27:56 +00:00
performance_timer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
performance_timer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
protobuf_utils.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
test_utils.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
unittest.proto
unpack.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
wav_based_simulator.cc Reland "Reland "Remove WEBRTC_TRACE."" 2017-10-04 14:40:44 +00:00
wav_based_simulator.h audioproc_f with simulated mic analog gain 2017-09-27 10:27:56 +00:00