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1beef1a97aa7e9a3f50776e100fa2bc7c63fcdd8
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7 Commits

Author SHA1 Message Date
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly 
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
 2018-07-05 10:59:49 +00:00
Steve Anton
d367921eb1 Configure media flow correctly with Unified Plan 
This also changes RtpReceiver and RemoteAudioSource to have two-step
initialization, since in Unified Plan RtpReceivers are created much
earlier than in Plan B.

Bug: webrtc:7600
Change-Id: Ia135d25eb8bcab22969007b3a825a5a43ce62bf4
Reviewed-on: https://webrtc-review.googlesource.com/39382
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21681}
 2018-01-18 19:01:38 +00:00
Steve Anton
6077675ab3 Change RtpReceivers to interact with the media channel directly 
Currently, the RtpReceivers take a BaseChannel which is (mostly)
just used for proxying calls to the MediaChannel. This change
removes the extra layer and moves the proxying logic to RtpReceiver.

Bug: webrtc:8587
Change-Id: I01b0e3d57b4629e43d9d148cc94d6dd2941d320e
Reviewed-on: https://webrtc-review.googlesource.com/38120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21562}
 2018-01-11 00:16:44 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros. 
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
 2017-11-09 11:56:32 +00:00
Steve Anton
3b80aace61 Fix flaky memory leak in RemoteAudioSource 
Bug: webrtc:8405
Change-Id: Ie7c89214323678c6ea34e344bb1a5a33ad46b3f0
Reviewed-on: https://webrtc-review.googlesource.com/13401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20362}
 2017-10-19 18:01:52 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/ 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
 2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/. 
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
 2017-09-15 04:25:06 +00:00