philipel 57ec685446 Unwrap TL0 pic index to avoid having to work with a wrapped number. ...

This is to avoid clearing the |gof_info_| map when there are jumps in the
tl0 pic index.

Bug: chromium:855211
Change-Id: I762557070d65b3c535cb9a49498975bcd9c2c485
Reviewed-on: https://webrtc-review.googlesource.com/86943
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23872}

2018-07-06 13:28:38 +00:00

..

audio_coding

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

audio_device

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

audio_mixer

Calculate all audio samples in AudioMixerCalculateEnergy.

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

audio_processing

Revert "IWYU: Add <cmath> for fabsf() and powf(), switch to C++ versions"

2018-07-06 11:37:15 +00:00

bitrate_controller

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

congestion_controller

Field trial to initialize throughput estimate faster.

2018-07-05 12:14:04 +00:00

desktop_capture

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

include

Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.

2018-07-05 14:29:07 +00:00

pacing

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

remote_bitrate_estimator

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

rtp_rtcp

Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.

2018-07-05 14:29:07 +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

Unwrap TL0 pic index to avoid having to work with a wrapped number.

2018-07-06 13:28:38 +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

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