webrtc/modules/audio_coding/acm2
Henrik Lundin 1ff41eb784 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"
This reverts commit 80c4cca491.

Reason for revert: Breaks downstream tests.

Original change's description:
> 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}

TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:44 +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 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-06-21 12:36:44 +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