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webrtc/modules/audio_coding/codecs
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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
..
cng Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g711 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g722 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
ilbc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
isac Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
opus Add field trials for configuring Opus encoder packet loss rate. 2018-10-15 08:59:43 +00:00
pcm16b Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
red Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
tools Delete root header file typedef.h. 2018-07-25 14:59:26 +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
audio_format_conversion.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_format_conversion.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
legacy_encoded_audio_frame.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00