Danil Chapovalov e40b437401 Discard frame self-dependency when parsing genric frame descriptor ...

Bug: chromium:859281
Change-Id: Ieb96f633a93f4f2e498bb1949339e239184bce9d
Reviewed-on: https://webrtc-review.googlesource.com/86545
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23817}

2018-07-03 10:28:05 +00:00

..

audio_coding

NetEq: Fold GetDecisionSpecialized into GetDecision

2018-07-02 14:51:09 +00:00

audio_device

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

audio_mixer

Calculate all audio samples in AudioMixerCalculateEnergy.

2018-06-29 14:47:13 +00:00

audio_processing

Fix fuzzer-found overflow in AGC1

2018-07-03 09:56:34 +00:00

bitrate_controller

Removes redundant delay based bwe.

2018-07-02 09:11:33 +00:00

congestion_controller

Removes redundant delay based bwe.

2018-07-02 09:11:33 +00:00

desktop_capture

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

include

Add RTPVideoHeader const accessor.

2018-06-21 09:49:40 +00:00

pacing

Removes redundant AlrDetector.

2018-06-29 16:28:04 +00:00

remote_bitrate_estimator

Removes redundant delay based bwe.

2018-07-02 09:11:33 +00:00

rtp_rtcp

Discard frame self-dependency when parsing genric frame descriptor

2018-07-03 10:28:05 +00:00

utility

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

video_capture

Delete unused file.

2018-06-28 12:53:17 +00:00

video_coding

Add VP9 profile negotiation to SDP

2018-07-02 23:38:41 +00:00

video_processing

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

BUILD.gn

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

module_common_types_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