Reason for revert:
Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6).
Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer.
Original issue's description:
> Enable GN check for webrtc/base
>
> It's not possible to enable it for the rtc_base_approved
> target but since a larger refactoring is ongoing for webrtc/base
> this CL doesn't attempt to fix that.
>
> Changes made:
> * Move webrtc/system_wrappers/include/stringize_macros.h into
> webrtc/base:rtc_base_approved_unittests (and corresponding
> unit test to rtc_base_approved_unittests).
> * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
> * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
> webrtc/base.
> * Remove unused use include of webrtc/base/fileutils.h in
> webrtc/base/pathutils.cc
>
> BUG=webrtc:6828, webrtc:3806, webrtc:7480
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2717083002
> Cr-Commit-Position: refs/heads/master@{#17766}
> Committed: ed754e71aeTBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6828, webrtc:3806, webrtc:7480
NOTRY=True
Review-Url: https://codereview.webrtc.org/2838683002
Cr-Commit-Position: refs/heads/master@{#17849}
The dependency on third_party/expat/ is removed.
The dependency on third_party/jsoncpp is removed from
libjingle_peerconnection while peerconnection_client still
depends on it.
BUG=webrtc:7516
Review-Url: https://codereview.webrtc.org/2832283002
Cr-Commit-Position: refs/heads/master@{#17848}
Once the buffer returned by Windows is not newly allocated, it may contain
legacy images from previous capturing attempts. This usually is not a problem,
as implementations other than ScreenCapturerWinDirectx paint each pixel on the
frame. But due to the one capturer per monitor design of
ScreenCapturerWinDirectx, part of the frame may not be covered by any
DxgiOutputDuplicator, and cause the legacy image to be shown.
So a very simple fix is to clear the DesktopFrame in DxgiFrame.
BUG=708766
Review-Url: https://codereview.webrtc.org/2827983007
Cr-Commit-Position: refs/heads/master@{#17847}
Reason for revert:
Breaks libfuzzer bot - you forgot to run that one :P
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%28Libfuzzer%29/builds/5571
Original issue's description:
> Removing test deps in webrtc/test/fuzzers
>
> Targets in webrtc/test/fuzzers are used in chromium which includes WebRTC
> with rtc_include_tests=false.
>
> We enabled 'gn check' on the webrtc/test directory and we have detected
> that some dependencies were not tracked. These dependencies are on test
> targets so we cannot add them in the dep list because this causes a
> breakage in chromium.
>
> BUG=webrtc:7515
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2840523003
> Cr-Commit-Position: refs/heads/master@{#17844}
> Committed: 14b86d3864TBR=mbonadei@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7515
Review-Url: https://codereview.webrtc.org/2835263002
Cr-Commit-Position: refs/heads/master@{#17846}
With this CL, all tests and tools under the neteq/ folder are
converted to use RTPHeader instead of WebRtcRTPHeader. WebRtcRTPHeader
has an RTPHeader as a member. None of the other member in
WebRtcRTPHeader where used.
TBR=kjellander@webrtc.org
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_compile_rel_ng,linux_chromium_compile_dbg_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2809153002
Cr-Commit-Position: refs/heads/master@{#17845}
Targets in webrtc/test/fuzzers are used in chromium which includes WebRTC
with rtc_include_tests=false.
We enabled 'gn check' on the webrtc/test directory and we have detected
that some dependencies were not tracked. These dependencies are on test
targets so we cannot add them in the dep list because this causes a
breakage in chromium.
BUG=webrtc:7515
NOTRY=True
Review-Url: https://codereview.webrtc.org/2840523003
Cr-Commit-Position: refs/heads/master@{#17844}
Reason for revert:
Downstream roadblock should be cleared by now. Relanding original patch.
Original issue's description:
> Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
>
> Reason for revert:
> Broke downstream dependencies.
>
> Original issue's description:
> > Change NetEq::InsertPacket to take an RTPHeader
> >
> > It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> > a member. None of the other member in WebRtcRTPHeader where used in
> > NetEq.
> >
> > This CL adapts the production code; tests and tools will be converted
> > in a follow-up CL.
> >
> > BUG=webrtc:7467
> >
> > Review-Url: https://codereview.webrtc.org/2807273004
> > Cr-Commit-Position: refs/heads/master@{#17652}
> > Committed: 4d027576a6
>
> TBR=ivoc@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2812933002
> Cr-Commit-Position: refs/heads/master@{#17657}
> Committed: 10d095d4f7R=ivoc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2835093002 .
Cr-Commit-Position: refs/heads/master@{#17843}
Also move RtpTransportControllerSendInterface to its own header file.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}
Update the AppRTCMobileTestStubbedVideoIO test to run on
phones without Internet connection. This is done by bringing up
a local instance of AppRTC on the Linux machine connected to
the Android device.
Running this test will need the webrtc.DEPS solution to be configured
for the checkout, since that will pull down the precompiled AppRTC
package that is needed.
Continued from http://crrev.com/2780493002#ps20001 (by kjellander@)
Continued from http://crrev.com/2741743002#ps180001 (by mandermo@)
BUG=webrtc:7185
Review-Url: https://codereview.webrtc.org/2825313002
Cr-Commit-Position: refs/heads/master@{#17838}
Reason for revert:
Relanding after fixing ARC issue.
Original issue's description:
> Revert of GN: Enable ARC for Mac and iOS in rtc_* templates (patchset #3 id:40001 of https://codereview.webrtc.org/2781713004/ )
>
> Reason for revert:
> Breaks mac build
>
> Original issue's description:
> > GN: Enable ARC for Mac and iOS in rtc_* templates
> >
> > Remove all uses of retain/release and NSAutoreleasePool.
> >
> > This makes transformation to Bazel easier.
> >
> > This CL subsumes https://codereview.webrtc.org/2778163002 and depends on https://codereview.webrtc.org/2784483002/
> >
> > BUG=webrtc:6412
> >
> > Review-Url: https://codereview.webrtc.org/2781713004
> > Cr-Commit-Position: refs/heads/master@{#17780}
> > Committed: 6bda02b51d
>
> TBR=kjellander@webrtc.org,magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6412
>
> Review-Url: https://codereview.webrtc.org/2827223003
> Cr-Commit-Position: refs/heads/master@{#17784}
> Committed: 7c8786ae8fTBR=kjellander@webrtc.org,magjed@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6412
Review-Url: https://codereview.webrtc.org/2834273002
Cr-Commit-Position: refs/heads/master@{#17836}
The signal was only being hooked up for incoming connections, not
outgoing connections.
As a result, the bandwidth estimator didn't know when packets were sent
and couldn't calculate delays.
BUG=webrtc:7509
Review-Url: https://codereview.webrtc.org/2834083002
Cr-Commit-Position: refs/heads/master@{#17817}
Otherwise, the activeCount will become negative.
BUG=webrtc:7471
Review-Url: https://codereview.webrtc.org/2822233002
Cr-Commit-Position: refs/heads/master@{#17816}
Introduce new small header-only targets in system_wrappers:
:cpu_features_api
:field_trial_api
:metrics_api
to untangle and optimize dependencies but still satisfy GN check.
In webrtc/p2p, previously uncovered header "base/fakecandidatepair.h"
is added to :p2p_test_utils target.
Refactor system_wrappers so 'rtc_p2p' can depend on only
system_wrappers:field_trial_api instead of all of system_wrappers
(which led to a breakage in Chromium that called for the revert of
https://codereview.webrtc.org/2735583002).
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2739863002
Cr-Commit-Position: refs/heads/master@{#17812}
Before this CL, we would negotiate:
- No crypto suites for data m= sections.
- A full set for audio m= sections.
- The full set, minus SRTP_AES128_CM_SHA1_32 for video m= sections.
However, this doesn't make sense with BUNDLE, since any DTLS
association could end up being used for any type of media. If
video is "bundled on" the audio transport (which is typical), it
will actually end up using SRTP_AES128_CM_SHA1_32.
So, this CL moves the responsibility of deciding SRTP crypto suites out
of BaseChannel and into DtlsTransport. The only two possibilities are
now "normal set" or "normal set + GCM", if enabled by the PC factory
options.
This fixes an issue (see linked bug) that was occurring when audio/video
were "bundled onto" the data transport. Since the data transport
wasn't negotiating any SRTP crypto suites, none were available to use
for audio/video, so the application would get black video/no audio.
This CL doesn't affect the SDES SRTP crypto suite negotiation;
it only affects the negotiation in the DLTS handshake, through
the use_srtp extension.
BUG=chromium:711243
Review-Url: https://codereview.webrtc.org/2815513012
Cr-Commit-Position: refs/heads/master@{#17810}
When SSRCs aren't signaled, an SSRC of 0 is used internally to mean
"the default receive stream". But this wasn't working with the
implementation of GetRtpReceiveParameters in the audio/video
engines. This was breaking RtpReceiver.GetParameters in this situation,
as well as the new getStats implementation (which relies on
GetParameters).
The new implementation will fail if *no* default receive stream is
configured (meaning no default sink is set), and otherwise will return
a default RtpEncodingParameters (later it will be filled with relevant
SDP parameters as they're implemented).
BUG=webrtc:6971
Review-Url: https://codereview.webrtc.org/2806173002
Cr-Commit-Position: refs/heads/master@{#17803}
