Jakob Ivarsson 83bd37cda4 Add field trials for configuring Opus encoder packet loss rate. ... Add options to: 1. Bypass optimization (use reported packet loss). 2. Set a maximum value. 3. Set a coefficient. Bug: webrtc:9866 Change-Id: I3fef43e5186a4f0f50fda3506e445860518cfbd7 Reviewed-on: https://webrtc-review.googlesource.com/c/105304 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25161}		2018-10-15 08:59:43 +00:00
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test	Moving LappedTransform, Blocker and AudioRingBuffer.	2018-08-31 15:27:50 +00:00
audio_decoder_opus.cc	Replace rtc::Optional with absl::optional in modules/audio_coding	2018-06-19 12:46:20 +00:00
audio_decoder_opus.h	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
audio_encoder_opus.cc	Add field trials for configuring Opus encoder packet loss rate.	2018-10-15 08:59:43 +00:00
audio_encoder_opus.h	Add field trials for configuring Opus encoder packet loss rate.	2018-10-15 08:59:43 +00:00
audio_encoder_opus_unittest.cc	Add field trials for configuring Opus encoder packet loss rate.	2018-10-15 08:59:43 +00:00
opus_bandwidth_unittest.cc	Moving LappedTransform, Blocker and AudioRingBuffer.	2018-08-31 15:27:50 +00:00
opus_complexity_unittest.cc	Optional: Use nullopt and implicit construction in /modules/audio_coding	2017-11-17 11:58:37 +00:00
opus_fec_test.cc	Enable clang::find_bad_constructs for audio_coding (part 1/2).	2018-07-20 13:07:47 +00:00
opus_inst.h	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
opus_interface.c	Implement Opus bandwidth adjustment behind a FieldTrial	2017-11-20 20:04:19 +00:00
opus_interface.h	Delete root header file typedef.h.	2018-07-25 14:59:26 +00:00
opus_speed_test.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00
opus_unittest.cc	Reformat the WebRTC code base	2018-06-19 14:00:39 +00:00