henrika 09a76193f5 Resolves threading issues when audio is interrupted on iOS. ...

Before this change we could crash in Debug when WebRTC audio was first
interrupted and then resumed again. The reason was that the new audio
stream stems from a new native I/O thread and that triggered thread
checkers. With this change, failing thread checkers are detached when
audio is interrupted to ensure that they don't fail when audio is restarted.

NOTRY=TRUE

Bug: webrtc:8126
Change-Id: Ib36ff6bc942477730aba60066f049ed0c43d3901
Reviewed-on: https://chromium-review.googlesource.com/628716
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19465}

2017-08-23 14:12:07 +00:00

..

audio_coding

Reland of Stop silently accepting unsupported flags in test binaries (patchset #1 id:1 of https://codereview.webrtc.org/3002963002/ )

2017-08-22 17:55:47 +00:00

audio_conference_mixer

Remove remains of webrtc/base

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

audio_device

Resolves threading issues when audio is interrupted on iOS.

2017-08-23 14:12:07 +00:00

audio_mixer

Remove remains of webrtc/base

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

audio_processing

Fix places that trigger no-unused-lambda-capture

2017-08-23 11:15:18 +00:00

bitrate_controller

Final version of BBR, with tweaks made for WebRTC, major changes:

2017-08-20 16:19:58 +00:00

congestion_controller

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

desktop_capture

Check child windows in CroppingWindowCapturerWin

2017-08-22 21:49:24 +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

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

remote_bitrate_estimator

Fix compile error for the win_msvc_rel bot.

2017-08-21 15:53:47 +00:00

rtp_rtcp

Reland of Add a flags field to video timing extension. (patchset #1 id:1 of https://codereview.webrtc.org/2995953002/ )

2017-08-18 09:51:12 +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 video send/receive stream headers to webrtc/call.

2017-08-22 12:43:23 +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