peah ea154106a8 Lowered the allowed jitter in the api calls to a reasonable level ...

This CL reduces the allowed jitter in the api calls to a reasonable
level in order to ensure a quicker revery from audio path glitches.

BUG=webrtc:8224, chromium:763775

Review-Url: https://codereview.webrtc.org/3009273002
Cr-Commit-Position: refs/heads/master@{#19772}

2017-09-11 13:44:37 +00:00

..

audio_coding

Implement ANA statistics.

2017-09-09 15:45:40 +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

Removes unused ADM APIs (final stage)

2017-09-11 13:17:38 +00:00

audio_mixer

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

audio_processing

Lowered the allowed jitter in the api calls to a reasonable level

2017-09-11 13:44:37 +00:00

bitrate_controller

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

congestion_controller

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

desktop_capture

Wrong coordinate is used in ScreenCapturerWinGdi

2017-09-07 17:42:34 +00:00

include

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

media_file

Reland "Supporting 48kHz PCM file."

2017-09-01 19:49:24 +00:00

pacing

Update thread annotiation macros in modules to use RTC_ prefix

2017-09-07 14:53:45 +00:00

remote_bitrate_estimator

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

rtp_rtcp

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

utility

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

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

video_capture

Decoupling video_capture_internal_imlp from Obj-C code

2017-09-04 13:54:54 +00:00

video_coding

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +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

Add VideoProcessorIntegrationTest for MediaCodec implementations.

2017-09-07 14:50:06 +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