peah cd8b079afb Increased the allowed API call jitter in AEC3 ...

This CL increases the amount of API call jitter that is allowed in AEC3
without causing resets of AEC3. This increase is now possible, as non-
causal alignments will be detected by the newly imposed delay bound.

BUG=8175

Review-Url: https://codereview.webrtc.org/3012553002
Cr-Commit-Position: refs/heads/master@{#19603}

2017-08-30 13:19:11 +00:00

..

audio_coding

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_conference_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_device

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_mixer

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

audio_processing

Increased the allowed API call jitter in AEC3

2017-08-30 13:19:11 +00:00

bitrate_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

congestion_controller

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

desktop_capture

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

include

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

media_file

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

pacing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

remote_bitrate_estimator

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

rtp_rtcp

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

utility

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_capture

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_coding

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

video_processing

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +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