sergeyu 6dbbd89f18 Fix BitrateProber to match the requested bitrate more precisely ...

Previously BirateProber was calculating delay between probes based on
the size of the previous probe. Because of that the actual sent bitrate
can deviate greatly from the target value. With this change it uses
total number of bytes in the cluster to estimate delay before each
probe.

BUG=webrtc:6952

Review-Url: https://codereview.webrtc.org/2613543003
Cr-Original-Commit-Position: refs/heads/master@{#15971}
Committed: 599c5011e7
Review-Url: https://codereview.webrtc.org/2613543003
Cr-Commit-Position: refs/heads/master@{#16127}

2017-01-17 23:07:59 +00:00

..

audio_coding

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +00:00

audio_conference_mixer

Refactor webrtc/modules/audio_conference_mixer for GN check

2016-12-22 14:56:21 +00:00

audio_device

Initial implementation of Android audio recording error handling.

2017-01-11 21:00:01 +00:00

audio_mixer

Delete voice_engine_configurations.h

2016-12-12 13:03:08 +00:00

audio_processing

GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.

2017-01-17 10:37:34 +00:00

bitrate_controller

Refactor bitrate_controller, remote_bitrate_estimator and congestion_contoller for gn check.

2016-12-28 12:43:46 +00:00

congestion_controller

Parse MedianSlopeFilter-parameters to the correct variables.

2017-01-17 11:30:05 +00:00

desktop_capture

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

include

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

media_file

Refactor webrtc/modules/media_file for GN check

2016-12-21 08:20:52 +00:00

pacing

Fix BitrateProber to match the requested bitrate more precisely

2017-01-17 23:07:59 +00:00

remote_bitrate_estimator

Reland of actor webrtc_perf_tests into several source_sets. (patchset #1 id:1 of https://codereview.webrtc.org/2613913002/ )

2017-01-05 15:09:50 +00:00

rtp_rtcp

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

utility

Move FilePlayer and FileRecorder to Voice Engine

2017-01-10 09:12:51 +00:00

video_capture

Replace ASSERT by RTC_DCHECK in all non-test code.

2017-01-12 13:15:36 +00:00

video_coding

Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )

2017-01-17 12:03:53 +00:00

video_processing

Reland "Refactor webrtc/modules/video_processing for GN check"

2016-12-27 14:32:08 +00:00

BUILD.gn

Added a new echo likelihood stat that reports the maximum value from a previous time period.

2017-01-15 16:29:46 +00:00

module_common_types_unittest.cc

Move functionality out from AudioFrame and into AudioFrameOperations.

2016-12-05 09:46:20 +00:00

OWNERS

Remove all references to GYP

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