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/opus
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
..
test Delete header file rtc_base/memory/aligned_array.h 2020-02-20 14:55:25 +00:00
audio_coder_opus_common.cc Decoder for multistream Opus. 2019-04-08 16:15:37 +00:00
audio_coder_opus_common.h Decoder for multistream Opus. 2019-04-08 16:15:37 +00:00
audio_decoder_multi_channel_opus_impl.cc WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
audio_decoder_multi_channel_opus_impl.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_decoder_multi_channel_opus_unittest.cc Fully qualify googletest symbols. 2019-04-09 17:18:20 +00:00
audio_decoder_opus.cc AudioDecoderOpus: Add support for 16 kHz output sample rate 2019-05-29 12:42:38 +00:00
audio_decoder_opus.h Format almost everything. 2019-07-08 13:45:15 +00:00
audio_encoder_multi_channel_opus_impl.cc Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
audio_encoder_multi_channel_opus_impl.h Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
audio_encoder_multi_channel_opus_unittest.cc Encoder side of Multistream Opus. 2019-04-25 15:07:38 +00:00
audio_encoder_opus.cc Revert "Default enable WebRTC-SendSideBwe-WithOverhead." 2020-10-22 16:57:18 +00:00
audio_encoder_opus.h opus: take SILK vad result into account for voice detection 2020-07-16 11:37:35 +00:00
audio_encoder_opus_unittest.cc Revert "Default enable WebRTC-SendSideBwe-WithOverhead." 2020-10-22 16:57:18 +00:00
opus_bandwidth_unittest.cc Update Opus tests for Opus 1.3 2020-03-05 08:53:37 +00:00
opus_complexity_unittest.cc Change gtest name to allow filtering based on the story name. 2020-09-11 14:11:27 +00:00
opus_fec_test.cc Make Opus PLC always output 10ms audio. 2019-11-07 21:15:58 +00:00
opus_inst.h Make Opus PLC always output 10ms audio. 2019-11-07 21:15:58 +00:00
opus_interface.cc opus: take SILK vad result into account for voice detection 2020-07-16 11:37:35 +00:00
opus_interface.h opus: add helper function to extract voice activity information 2020-06-10 14:21:01 +00:00
opus_speed_test.cc WebRTC Opus C interface: Add support for non-48 kHz decode sample rate 2019-05-29 10:33:03 +00:00
opus_unittest.cc opus: take SILK vad result into account for voice detection 2020-07-16 11:37:35 +00:00