webrtc/modules/audio_coding/acm2
Henrik Lundin 80c4cca491 NetEq: Deprecate playout modes Fax, Off and Streaming
The playout modes other than Normal have not been reachable for a long
time, other than through tests. It is time to deprecate them.

The only meaningful use was that Fax mode was sometimes set from
tests, in order to avoid time-stretching operations (accelerate and
pre-emptive expand) from messing with the test results. With this CL,
a new config is added instead, which lets the user specify exactly
this: don't do time-stretching.

As a result of Fax and Off modes being removed, the following code
clean-up was done:
- Fold DecisionLogicNormal into DecisionLogic.
- Remove AudioRepetition and AlternativePlc operations, since they can
  no longer be reached.

Bug: webrtc:9421
Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
Reviewed-on: https://webrtc-review.googlesource.com/84123
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23704}
2018-06-21 11:51:21 +00:00
..
acm_codec_database.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_codec_database.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receive_test.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
acm_receive_test.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_receiver.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
acm_receiver_unittest.cc NetEq: Deprecate playout modes Fax, Off and Streaming 2018-06-21 11:51:21 +00:00
acm_resampler.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_resampler.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
acm_send_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
acm_send_test.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
audio_coding_module.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_coding_module_unittest.cc Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. 2018-06-19 16:44:19 +00:00
call_statistics.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
call_statistics.h Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
call_statistics_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
codec_manager.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
codec_manager.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
codec_manager_unittest.cc Removed Die mock from MockAudioEncoder 2018-02-22 12:53:38 +00:00
rent_a_codec.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
rent_a_codec_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00