mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 13:50:40 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_coding/codecs
History
Jakob Ivarsson 27af3c4c24 Revert "Default enable WebRTC-SendSideBwe-WithOverhead." 
This reverts commit 87c1950841.

Reason for revert: breaks downstream tests

Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}

TBR=stefan@webrtc.org,jakobi@webrtc.org,alito@webrtc.org

Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
..
cng Rename several more tests that use EXPECT_DEATH to *DeathTest. 2020-05-18 16:10:04 +00:00
g711 Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
g722 Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
ilbc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE. 2020-05-04 15:01:26 +00:00
isac Revert "Default enable WebRTC-SendSideBwe-WithOverhead." 2020-10-22 16:57:18 +00:00
opus Revert "Default enable WebRTC-SendSideBwe-WithOverhead." 2020-10-22 16:57:18 +00:00
pcm16b Format almost everything. 2019-07-08 13:45:15 +00:00
red red: do not generate packets which are > 1200 bytes 2020-08-04 09:53:47 +00:00
tools Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 2019-08-07 13:36:05 +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
builtin_audio_decoder_factory_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
builtin_audio_encoder_factory_unittest.cc Format almost everything. 2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame.cc [Cleanup] Add missing #include. Remove useless ones. 2018-10-23 11:32:56 +00:00
legacy_encoded_audio_frame.h Format almost everything. 2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame_unittest.cc Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00