Philipp Hancke 686a3709ac opus: take SILK vad result into account for voice detection ... BUG=webrtc:11643 Change-Id: Idc3a9b6bb7bd1a33f905843e5d6067ae19d5172c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176508 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31743}		2020-07-16 11:37:35 +00:00
..
cng	Rename several more tests that use EXPECT_DEATH to *DeathTest.	2020-05-18 16:10:04 +00:00
g711	Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.	2020-03-25 22:19:21 +00:00
g722	Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.	2020-03-25 22:19:21 +00:00
ilbc	Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.	2020-05-04 15:01:26 +00:00
isac	iSAC encoder: Make it possible to change target bitrate at any time	2020-06-22 14:59:22 +00:00
opus	opus: take SILK vad result into account for voice detection	2020-07-16 11:37:35 +00:00
pcm16b	Format almost everything.	2019-07-08 13:45:15 +00:00
red	red: implement RED with distance 2	2020-07-03 13:53:28 +00:00
tools	Add RTC_ prefix to non-standard format specifier macro "PRIdNS"	2019-08-07 13:36:05 +00:00
audio_decoder.h	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
audio_encoder.h	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
builtin_audio_decoder_factory_unittest.cc	Format almost everything.	2019-07-08 13:45:15 +00:00
builtin_audio_encoder_factory_unittest.cc	Format almost everything.	2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
legacy_encoded_audio_frame.h	Format almost everything.	2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame_unittest.cc	Stop using Googletest legacy APIs.	2019-01-31 13:23:33 +00:00