asapersson 142fcc96d6 Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings. ...

Make it possible for forced VP8 SW fallback encoder to set min_pixels_per_frame via GetScalingSettings().

Add a min required resolution (in addition to bitrate) before releasing forced SW fallback.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3000693003
Cr-Commit-Position: refs/heads/master@{#19390}

2017-08-17 15:58:54 +00:00

..

audio_coding

Make isac_fix_test use gtest (in a hacky way)

2017-08-17 15:25:28 +00:00

audio_conference_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_device

Avoids WebRtcAudioTrack null pointer access at stop.

2017-08-16 13:14:08 +00:00

audio_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_processing

Replace absolute path with relative path for GN files.

2017-08-15 15:57:36 +00:00

bitrate_controller

Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

2017-08-09 13:13:45 +00:00

congestion_controller

Reland of Make the acceptable queue in the cwnd experiment configurable. (patchset #1 id:1 of https://codereview.webrtc.org/2999893002/ )

2017-08-17 09:13:54 +00:00

desktop_capture

Replace WindowUnderPoint free function with WindowFinder interface

2017-08-16 17:45:18 +00:00

include

Add AudioFrame::ResetWithoutMuting() to address performance regression.

2017-07-08 23:36:26 +00:00

media_file

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

pacing

Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )

2017-08-16 15:16:25 +00:00

remote_bitrate_estimator

Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ )

2017-08-16 15:16:25 +00:00

rtp_rtcp

When using clang, switch on -Wc++11-narrowing

2017-08-16 11:52:35 +00:00

utility

Tracking mock_process_thread with a GN target

2017-08-03 12:57:11 +00:00

video_capture

Replace old logging mechanism in webrtc/modules/video_capture/linux

2017-08-16 13:51:27 +00:00

video_coding

Move kMinPixelsPerFrame constant in VideoStreamEncoder to VideoEncoder::ScalingSettings.

2017-08-17 15:58:54 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00