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

..

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_unittest.cc

Add field trials for configuring Opus encoder packet loss rate.

2018-10-15 08:59:43 +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

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