Move it to include path of the rtp_rtcp module to indicate it is ok to include it outside of the module.
Renamed to match the class it introduce and to reduce confusion with rtp_header_extensions.h
Bug: webrtc:5565
Change-Id: Ic4b4e9f6b75cb9275e23539cd6e88632c1e7c8d2
Reviewed-on: https://chromium-review.googlesource.com/520947
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18402}
Prior to this CL, the ForwardErrorCorrection state would be reset whenever
the difference in sequence numbers of the last recovered media packet
and the new packet (media or FEC) was too large. This comparison did not
take into account that FlexFEC uses a different SSRC for the FEC packets,
meaning that the the state would be reset very frequently when FlexFEC
is used. This should not have led to any major problems, except for a
decreased decoding efficiency.
This CL verifies that whenever we compare sequence numbers in
ForwardErrorCorrection, they do indeed belong to the same SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2893293003
Cr-Commit-Position: refs/heads/master@{#18399}
Before 10.12, OSX may report 1X cursor on Retina screen. (See crbug.com/632995.)
After 10.12, OSX may report 2X cursor on non-Retina screen. (See
crbug.com/671436.) So scaling the cursor if the image size doesn't meet the
expected size on either Retina or non-Retina screen.
Also corrects the cursor caching and change detection, so we can only do scalingat cursor changing for better performance.
As to screen capture on OSX, the captured frame already contains the current
cursor. So the MouseCursorMonitorMac is not needed for ScreenCapture for
performance purpose.
BUG=671436
Review-Url: https://codereview.webrtc.org/2908853002
Cr-Commit-Position: refs/heads/master@{#18393}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
A recent change (https://codereview.webrtc.org/2855143002/) introduced
a bug in RtcEventLogSource::NextPacket(). The rtp_packet_index_ must
be incremented when a valid packet is found and delivered. Otherwise,
the same packet will be delivered over and over again.
The recent change also altered the way that audio packets are sifted out. Now, the RTP header is always parsed before discarding any non-audio packets. This means that RtpHeaderParser::Parse is always called, also with video packets, which sometimes contain padding. When header-only dumps (such as RtcEventLogs) are created, the payload is stripped, and the payload length is equal to
the RTP header length. However, if the original packet was padded, the
RTP header will carry information about this padding length, and the
parser will check that the pyaload length is at least the header +
padding. This is not the case for header-only dumps, and the parser will return an error. In this CL, we ignore that error when a header-only packet has padding length larger than 0.
BUG=webrtc:7538
Review-Url: https://codereview.webrtc.org/2912323003
Cr-Commit-Position: refs/heads/master@{#18385}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
Will also ensure that full-duplex audio now works on iOS simulators.
Bug: b/37580746
Change-Id: Iab1af39b0e6e6c124435814558caf77c474bd612
Reviewed-on: https://chromium-review.googlesource.com/519246
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18381}
ScheduledExecutorService silently ignores exceptions thrown by the
runnable. This makes debugging issues unnecessarily difficult.
Bug: None
Change-Id: I7deb43b96e5639c096b9aed9c6ff9b197b62f59f
Reviewed-on: https://chromium-review.googlesource.com/521084
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18378}
Decision was made to keep this API for the time being.
Bug: webrtc:7710
Change-Id: Ief41ffb2ec2345e3a74fc72927d038be1ff5941c
No-Try: True
Reviewed-on: https://chromium-review.googlesource.com/521085
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18377}
Reason for revert:
Fixing and re-landing.
Original issue's description:
> Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
>
> Reason for revert:
> ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
> #include "webrtc/base/checks.h"
> ^-------------------
> The target:
> //webrtc/test:fileutils_unittests
> is including a file from the target:
> //webrtc/base:rtc_base_approved
>
> It's usually best to depend directly on the destination target.
> In some cases, the destination target is considered a subcomponent
> of an intermediate target. In this case, the intermediate target
> should depend publicly on the destination to forward the ability
> to include headers.
>
> Dependency chain (there may also be others):
> //webrtc/test:fileutils_unittests -->
> //webrtc/test:fileutils --[private]-->
> //webrtc/base:rtc_base_approved
>
>
> Original issue's description:
> > Enabling `gn check` on webrtc/test
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2911203002
> > Cr-Commit-Position: refs/heads/master@{#18372}
> > Committed: db5bb404b0
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2920763002
> Cr-Commit-Position: refs/heads/master@{#18375}
> Committed: 1a6f143d07TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2918793002
Cr-Commit-Position: refs/heads/master@{#18376}
Reason for revert:
ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
#include "webrtc/base/checks.h"
^-------------------
The target:
//webrtc/test:fileutils_unittests
is including a file from the target:
//webrtc/base:rtc_base_approved
It's usually best to depend directly on the destination target.
In some cases, the destination target is considered a subcomponent
of an intermediate target. In this case, the intermediate target
should depend publicly on the destination to forward the ability
to include headers.
Dependency chain (there may also be others):
//webrtc/test:fileutils_unittests -->
//webrtc/test:fileutils --[private]-->
//webrtc/base:rtc_base_approved
Original issue's description:
> Enabling `gn check` on webrtc/test
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2911203002
> Cr-Commit-Position: refs/heads/master@{#18372}
> Committed: db5bb404b0TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2920763002
Cr-Commit-Position: refs/heads/master@{#18375}
Constants kRTCVp8CodecName, kRTCVp9CodecName, kRTCH264CodecName declared
in RTCRtpCodecParameters.h were defined without the "RTC" part in their
names, thus using them in the app code resulted in linking errors. This
patch fixes the naming mismatch.
BUG=webrtc:7721
Review-Url: https://codereview.webrtc.org/2910863002
Cr-Commit-Position: refs/heads/master@{#18363}
Similar to the existing constants for the media constraint-keys intended
for media sources, add the constants for the standard keys to generate
offers and answers.
This patch also adds a few comments to RTCMediaConstraints.h, to give
a better clue on the intended usage scope of declared media-constraint
keys and values.
BUG=webrtc:7722
Review-Url: https://codereview.webrtc.org/2908013002
Cr-Commit-Position: refs/heads/master@{#18362}
Only will be used if WEBRTC_POSIX and WEBRTC_LINUX are both defined and
"epoll_create" doesn't return an error. Otherwise the default "select"-based
IO loop will be used.
BUG=webrtc:7585
Review-Url: https://codereview.webrtc.org/2880923002
Cr-Commit-Position: refs/heads/master@{#18359}
1. Create unit tests for RtpDemuxer.
2. Add an RTC_DCHECK in RtpDemuxer that makes sure that the sink<->ssrc multimap does not allow multiple instances of the same association.
BUG=None
Review-Url: https://codereview.webrtc.org/2902823004
Cr-Commit-Position: refs/heads/master@{#18357}
Delete unused member |rtp_receiver_|, and simplify a return statement.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2912363002
Cr-Commit-Position: refs/heads/master@{#18354}
Use an overloaded version of UpdateNetworkMetrics() which does not require us to explicitly pass in an rtc::Optional.
BUG=None
Review-Url: https://codereview.webrtc.org/2899313004
Cr-Commit-Position: refs/heads/master@{#18347}
The old video send stream configs could contain multiple SSRCs and RTX SSRCs (in case of simulcast). To ensure that the RtcEventLog parser is backwards compatible, we have to return one config for every SSRC.
Also update the parsing functions for the other config types to return the config instead of passing in an output parameter.
BUG=webrtc:7731
Review-Url: https://codereview.webrtc.org/2912113002
Cr-Commit-Position: refs/heads/master@{#18343}
Also deletes a couple of includes of call.h, which seem
unnecessary.
BUG=None
Review-Url: https://codereview.webrtc.org/2907403003
Cr-Commit-Position: refs/heads/master@{#18340}
(This CL concerns both the PLR-based as well as the RPLR-based versions of FecController.)
1. Make FecController disable only when below the disabling-threshold, so as to prevent toggling when the enabling-curve and the disabling-curve are identical.
2. Extend unit-test coverage accordingly.
BUG=None
Review-Url: https://codereview.webrtc.org/2906663002
Cr-Commit-Position: refs/heads/master@{#18337}
Add test for vp8/vp9 qp parser in both videoprocessor_integrationtest.
Check the qp from parser equal to that from the encoder
on every frame in every test.
Add test for vp8/vp9 qp parser in vp8/vp9_impl_test.
Check the qp parser on a single key frame.
BUG=None
Review-Url: https://codereview.webrtc.org/2903163002
Cr-Commit-Position: refs/heads/master@{#18334}
The previous limit leaved no margin for RTT.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2911243002
Cr-Commit-Position: refs/heads/master@{#18333}
