Sebastian Jansson f34116e356 Replacing bandwidth adaptation trial with stable target in Opus encoder. ... This also means that the NetworkEstimate::bandwidth can be deprecated as it's currently just a copy of the target_rate. Bug: webrtc:10981 Change-Id: I1bc57b98480bd77ce052736b19d630c775428546 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29288}		2019-09-24 16:35:02 +00:00
..
cng	Use std::make_unique instead of absl::make_unique.	2019-09-17 15:47:29 +00:00
g711	Format almost everything.	2019-07-08 13:45:15 +00:00
g722	Format almost everything.	2019-07-08 13:45:15 +00:00
ilbc	Guard against clang-format wrong behavior.	2019-07-16 12:18:32 +00:00
isac	Delete AudioDecoder method IncomingPacket	2019-09-24 08:30:24 +00:00
opus	Replacing bandwidth adaptation trial with stable target in Opus encoder.	2019-09-24 16:35:02 +00:00
pcm16b	Format almost everything.	2019-07-08 13:45:15 +00:00
red	Format almost everything.	2019-07-08 13:45:15 +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