Rasmus Brandt 7b92ceb0ee Ensure that input_frames_.size() <= kMaxBufferedInputFrames at enqueue time. ... Bug: webrtc:9452 Change-Id: I6d415a2cb24461d7359ff30e6499d65d88d2b2f8 Reviewed-on: https://webrtc-review.googlesource.com/85371 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23846}		2018-07-05 07:08:59 +00:00
..
audio_coding	NetEq: Handle nested RED packets	2018-07-03 20:27:57 +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	AEC3: Reverberation model: Changes on the decay estimation.	2018-07-04 10:04:32 +00:00
bitrate_controller	Removes redundant delay based bwe.	2018-07-02 09:11:33 +00:00
congestion_controller	Adds debug printing for congestion controllers.	2018-07-03 17:00:24 +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	Add ParsedPayload::video_header() accessor.	2018-07-04 09:31:21 +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	Ensure that input_frames_.size() <= kMaxBufferedInputFrames at enqueue time.	2018-07-05 07:08:59 +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