Alessio Bazzica 330bf4076e WebRTC VAD wrapper for APM-QA ...

Alternative VAD based on the existing one in WebRTC.
It is used to extract VAD annotations in APM-QA.

TBR=

Bug: webrtc:7494
Change-Id: I6af412742f804631ad4f3ba3ccf71a30d74de984
Reviewed-on: https://webrtc-review.googlesource.com/14553
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20404}

2017-10-24 08:34:38 +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

WebRTC VAD wrapper for APM-QA

2017-10-24 08:34:38 +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_unittest.cc

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +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

fake_recording_device_unittest.cc

audioproc_f with simulated mic analog gain

2017-09-27 10:27:56 +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

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