The video send pipeline uses the magic value 0 for an unset ntp time.
However, the receive pipeline uses the magic value -1 for unset
(unclear where, it seems it behaved differently a few months ago).
This makes cl https://codereview.webrtc.org/2469993003/ fail the
P2PTestConductor.ForwardVideoOnlyStream, because that cl removes code
which always clears the ntp time, and enables propagation of ntp time
from the receive pipeline to the send pipeline.
Treating ntp time <= 0 as unset is a small improvement. Ultimately, a
VideoFrame shouldn't carry an ntp time at all.
BUG=webrtc:5740,webrtc:6977
Review-Url: https://codereview.webrtc.org/2620383005
Cr-Commit-Position: refs/heads/master@{#16035}
We currently leak one local reference to MediaCodecVideoEncoder in
every call to MediaCodecVideoEncoderFactory::CreateVideoEncoder. After
the encoder has been re-initialized 512 times, JNI will crash due to
local reference table overflow (max=512).
The actual leak is in the member initializer list of
MediaCodecVideoEncoder. This CL fixes the leak by adding a
ScopedLocalRefFrame outside of the ctor. All JNI code that originate
from a C++ thread (i.e. the entry point is not a Java thread) must use
a ScopedLocalRefFrame in order to avoid leaking local references.
BUG=webrtc:6969,b/34056152
Review-Url: https://codereview.webrtc.org/2627973004
Cr-Commit-Position: refs/heads/master@{#16034}
This is a slightly more descriptive name, since we only have one type
of erasure code (XOR), and we only have one table.
BUG=None
Review-Url: https://codereview.webrtc.org/2625903004
Cr-Commit-Position: refs/heads/master@{#16032}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
This change moves the definition of the UMA MetricsObserverInterface from api/peerconnectioninterface.h into api/umametrics.h. This allows us to remove the unwanted dependency on peerconnectioninterface.h from files in webrtc/p2p.
This is a simple refactoring with no functional changes.
BUG=None
Review-Url: https://codereview.webrtc.org/2627093005
Cr-Commit-Position: refs/heads/master@{#16020}
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.
Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Original-Commit-Position: refs/heads/master@{#15777}
Committed: 7a5fa6cd61
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#16016}
Trivial change that allows users to call MicrophoneVolumeIsAvailable()
(and get a valid result) on Android without crashing.
TBR=henrik.lundin
BUG=NONE
Review-Url: https://codereview.webrtc.org/2620243003
Cr-Commit-Position: refs/heads/master@{#16013}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
Deps have rolled to 1.6, and since no one noticed that the old code path
was broken and wouldn't even compile, I assume no one is using it.
I therefore deem it time to clean away all these nasty ifdefs.
("const kNalHeaderSizeAllocation = 50;" doesn't declare a type)
BUG=chromium:614970
Review-Url: https://codereview.webrtc.org/2622233002
Cr-Commit-Position: refs/heads/master@{#16008}
Avoids confusion about the meaning of "incoming".
BUG=webrtc:6897
Review-Url: https://codereview.webrtc.org/2624073003
Cr-Commit-Position: refs/heads/master@{#16007}
In this CL:
- Removed unused variable |last_seq_num_|.
- Fixed bug where a new incomplete frame was detected as a complete frame.
- Added fuzzer to video_coding::PacketBuffer.
BUG=chromium:677101
Review-Url: https://codereview.webrtc.org/2613833003
Cr-Commit-Position: refs/heads/master@{#16003}
Also enable Intel HW Vp8 encoder by default in AppRTCMobile.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2614373004
Cr-Commit-Position: refs/heads/master@{#16002}
This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.
Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.
BUG=wertc:6847
Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}
The purpose is to be able to add field trials in Java code.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2621003002
Cr-Commit-Position: refs/heads/master@{#15994}
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.
BUG=webrtc:5880
Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
Since packets can be received out of order NACKs could be triggered before any
packet was actually dropped. This cause the test to never drop packets which in
turn caused the block of code which set the |observation_complete_| event to
never execute.
BUG=webrtc:2845
Review-Url: https://codereview.webrtc.org/2613443002
Cr-Commit-Position: refs/heads/master@{#15990}
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@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/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
