gtest can print objects if they have an operator<< or a PrintTo
function in the same namespace as the object's class. Since
std::optional does not seem to have an operator<<, it'd be preferable
not to rely on rtc::Optional being printable through operator<<.
Currently, gtest errors will just dump the raw bytes of
rtc::Optionals, which make them really annoying to work with in tests.
BUG=webrtc:7196
Review-Url: https://codereview.webrtc.org/2704483002
Cr-Commit-Position: refs/heads/master@{#16717}
This CL adds the ability to _create_ HW codecs (Android and iOS) in the
VideoProcessor integration tests. Since the VideoProcessor class is not thread
safe yet, this CL does not add the ability to _use_ HW codecs in the tests. A
follow-up CL is planned that will add this ability.
This CL further adds a separate build target which is used to separate the
"plot" versions of the integration tests from the "correctness" versions. The
former will be run manually on devices, whereas the latter are used on the
trybots/buildbots to find regressions in the SW codecs. The underlying test
is the same, however.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2695653002
Cr-Commit-Position: refs/heads/master@{#16716}
Running video loopback on https://appr.tc/ revealed that it is possible
to use the same SSRC for a local and remote audio or video track. This
caused a DCHECK crash. The constructor of TrackMediaInfoMap is updated
to support this mapping and the unittest is updated (moved and modified
a test from being a death test to being a non-death test).
I've verified that this fixes the bug.
BUG=chromium:693087
Review-Url: https://codereview.webrtc.org/2703783002
Cr-Commit-Position: refs/heads/master@{#16713}
This is step 1 in the following process to move the task runner
abstraction over to Chrome, without gettings link errors on duplicate
symbols.
1. Move files from the rtc_base target to a new target
rtc_task_runner, and let rtc_base publicly depend on it.
2. In Chrome, add an explicit dependency on rtc_task_runner where it
depends on rtc_base.
3. Drop the webrtc dependency rtc_base --> rtc_task_runner.
4. Copy task runner code to Chrome (cl
https://codereview.chromium.org/2694903005/), and drop its
dependency on webrtc's rtc_task_runner target.
5. Delete the rtc_task_runner target and corresponding source files
from webrtc. Mission accomplished!
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2696703009
Cr-Commit-Position: refs/heads/master@{#16710}
This reduces binary size considerably and solves some other problems.
Also rewrote using variadic templates.
Initial patch contributed by andrey.semashev@gmail.com.
BUG=webrtc:2305
Review-Url: https://codereview.webrtc.org/2509733003
Cr-Commit-Position: refs/heads/master@{#16703}
This change adds a flag, use_sctpmap, to DataContentDescription. The deserialization code sets the flag based on the format of the m= line.
There were already unit tests using SDP in the new format, so I just updated them to check use_sctpmap was set as expected.
The change to mediasession copies use_sctpmap from the offered DataContentDescription to the answer.
I haven't figured out how to test this change yet, but wanted to get feedback before continuing.
BUG=chromium:686212
Review-Url: https://codereview.webrtc.org/2690943011
Cr-Commit-Position: refs/heads/master@{#16686}
The AsyncClosures only ever have one thing referencing them, so they
should be using std::unique_ptr to manage ownership. Maybe this code was
written before std::unique_ptr was available.
Originally reverted because it made a change to ScopedMessageData
that wasn't backwards compatible, and applications using the rtc::Thread
infrastructure may be using it.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2689233003
Cr-Commit-Position: refs/heads/master@{#16684}
Reason for revert:
The change to messagequeue.h isn't backwards compatible. Will reland after making it backwards compatible.
Original issue's description:
> Use std::unique_ptr instead of rtc::scoped_refptr in AsyncInvoker.
>
> The AsyncClosures only ever have one thing referencing them, so they
> should be using std::unique_ptr to manage ownership. Maybe this code was
> written before std::unique_ptr was available.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2689233003
> Cr-Commit-Position: refs/heads/master@{#16680}
> Committed: a5a472927bTBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2703613006
Cr-Commit-Position: refs/heads/master@{#16683}
The first unsignalled SSRC creates a default receive channel.
Any unsignalled SSRC changes after that replace the default SSRC.
Add unit tests for changing unsignalled SSRCs.
BUG=webrtc:5208
Review-Url: https://codereview.webrtc.org/2692993009
Cr-Commit-Position: refs/heads/master@{#16682}
The AsyncClosures only ever have one thing referencing them, so they
should be using std::unique_ptr to manage ownership. Maybe this code was
written before std::unique_ptr was available.
BUG=None
Review-Url: https://codereview.webrtc.org/2689233003
Cr-Commit-Position: refs/heads/master@{#16680}
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.
Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.
BUG=webrtc:7183
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
FallbackDesktopCapturerWrapper is a DesktopCapturer implementation, which owns
two DesktopCapturer implementations. If the main DesktopCapturer fails, it uses
the secondary capturer. The logic is now used in ScreenCapturerWinMagnifier, and
it can also be shared in ScreenCapturerWinDirectx to fallback to Gdi capturer on
privilege prompt or login screen.
BUG=684937
Review-Url: https://codereview.webrtc.org/2697453002
Cr-Commit-Position: refs/heads/master@{#16677}
This does not fix the myriad of other problems here, but at least
removes the dependency on CONF_VALUE.
BUG=526270
Review-Url: https://codereview.webrtc.org/2705603003
Cr-Commit-Position: refs/heads/master@{#16676}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16675}
This is necessary in case the drawer doesn't cover all the pixels.
BUG=None
Review-Url: https://codereview.webrtc.org/2704663002
Cr-Commit-Position: refs/heads/master@{#16671}
and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
