A handful of helpers were using SessionDescriptionInterface** output
arguments to return ownership. Chenge them to either use a
rtc::scoped_ptr<SessionDescriptionInterface>* output parameter, or to
simply return a rtc::scoped_ptr<SessionDescriptionInterface>. Not
using raw pointers for things you own is good in general; it will also
be very convenient when scoped_ptr is gone, since unique_ptr doesn't
have .accept() or .use() methods.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1798173002
Cr-Commit-Position: refs/heads/master@{#12021}
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.
The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.
BUG=webrtc:5636
Review URL: https://codereview.webrtc.org/1793553002
Cr-Commit-Position: refs/heads/master@{#12019}
Reason for revert:
Caused unexpected perf stats changes, see http://crbug/594575.
Original issue's description:
> VCMCodecTimer: Change filter from max to 95th percentile
>
> The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
>
> BUG=b/27306053
>
> Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> Cr-Commit-Position: refs/heads/master@{#11952}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1808693002
Cr-Commit-Position: refs/heads/master@{#12018}
Reason for revert:
Revert because it breaks downstream code.
Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1812453002
Cr-Commit-Position: refs/heads/master@{#12016}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
Part 3 of refactor. Also:
- better weak pointer delegate storage + tests
- we now ignore route changes when we're interrupted
- fixed bug where preferred sample rate wasn't set if audio session
wasn't active
BUG=
Review URL: https://codereview.webrtc.org/1796983004
Cr-Commit-Position: refs/heads/master@{#12007}
Instead of using a raw pointer output parameter. This is a good idea
in general, but will also be very convenient when scoped_ptr is gone,
since unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1799233002
Cr-Commit-Position: refs/heads/master@{#12004}
This is a good idea in general, because it makes ownership clearer,
but will also be very convenient when scoped_ptr is gone, since
unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1800753003
Cr-Commit-Position: refs/heads/master@{#12002}
Add WEBRTC_HAS_NEON and __SSE2__, otherwise, CPU detection fails in some cases.
Review URL: https://codereview.webrtc.org/1784323005
Cr-Commit-Position: refs/heads/master@{#12001}
this class replace and extend RTCPUtility::RtcpCommonHeader structure and RTCPUtility::RtcpParseCommonHeader function.
In addition to header fields, payload pointer is stored because rtcp header without payload is rarely useful.
Sample usage can be checked in 'RTCP Parser sketched' CL: https://codereview.webrtc.org/1555683002/
BUG=webrtc:5260
R=asapersson@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1575413002 .
Cr-Commit-Position: refs/heads/master@{#11999}
This CL removes the dependency of AudioProcessing in
EchoCancellerImpl. It is breaking the public APM API by
having a different error code behavior so please review it
carefully. I made a comment about the API breaking change
in the code section of this CL.
BUG=webrtc:5337
Review URL: https://codereview.webrtc.org/1770823002
Cr-Commit-Position: refs/heads/master@{#11998}
This CL adds the width and height to the createPbufferSurface exception
message. Also, when a Callable.call() fails in
ThreadUtils.invokeUninterruptibly() we rethrow a new exception, but that
excludes the callstack from Callable.call(). This CL adds the callstack
from Callable.call() to make debugging easier.
BUG=b/27581640,b/27516991
Review URL: https://codereview.webrtc.org/1780183005
Cr-Commit-Position: refs/heads/master@{#11996}
"WebRTC.Video.AVSyncOffsetInMs"
The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.
Updated sync tests in call_perf_tests.cc to use this implementation.
BUG=webrtc:5493
Review URL: https://codereview.webrtc.org/1756193005
Cr-Commit-Position: refs/heads/master@{#11993}
Devices exist (specifically the 16GB 5th generation iPod Touch) that do not have
a rear-facing camera.
This CL:
- Adjusts RTCAVFoundationVideoCapturerInternal so initialization doesn't fail
because a rear-facing camera doesn't exist, but still logs a warning
- Provides a check for whether or not a rear-facing camera can be used (useful
for toggling UI elements)
- Turns an attempt to switch to the rear-facing camera into a no-op with a
warning.
BUG=
Review URL: https://codereview.webrtc.org/1416303003
Cr-Commit-Position: refs/heads/master@{#11992}
Greatly speeds up setRemoteDescription() by moving encoder initialization
off the main worker thread, which is free to move onto gathering ICE
candidates and other tasks while InitEncode() is performed. It also
un-blocks PeerConnection GetStats() which is no longer blocked on
encoder initialization.
BUG=webrtc:5410
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1757313002 .
Cr-Commit-Position: refs/heads/master@{#11983}
In this CL:
- Added more general version of ForwardDiff/RevereseDiff.
- Replaced macro for testing if type T is of unsiged int with
std::is_unsigned<T> instead.
- Changed from EXPECT_XXX to ASSERT_XXX in the unittests since a lot of
the test contains loops that would otherwise just spam the terminal.
BUG=
Review URL: https://codereview.webrtc.org/1786043004
Cr-Commit-Position: refs/heads/master@{#11982}
- Change argument type to int for SetSendTelephoneEventPayloadType()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1798903002
Cr-Commit-Position: refs/heads/master@{#11980}
This CL changes the interface by adding a SurfaceTextureHelper argument
to VideoCapturer.startCapture(). This removes the need for the
VideoCapturer to create the SurfaceTextureHelper itself. This also means
that it is no longer necessary to send an EGLContext to the
VideoCapturerAndroid.create() function.
The SurfaceTextureHelper is now created in AndroidVideoCapturerJni, and
the EGLContext is passed from PeerConnectionFactory in
nativeCreateVideoSource().
Another change in this CL is that the C++ SurfaceTextureHelper creates
the Java SurfaceTextureHelper instead of getting it passed as an
argument in the ctor.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1783793002
Cr-Commit-Position: refs/heads/master@{#11977}
- Clean up unused methods in voe::Channel following removal of VoEDtmf APIs.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1785643006
Cr-Commit-Position: refs/heads/master@{#11976}
In addition:
- Introduces RTCAudioSessionTest
- iOS/Mac gtests now have an autoreleasepool
- Moves ScopedAutoreleasePool to rtc_base_approved
- Introduces route change button in AppRTCDemo
BUG=webrtc:5649
Review URL: https://codereview.webrtc.org/1782363002
Cr-Commit-Position: refs/heads/master@{#11971}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
Updated peerconnection_server to not need stuff from rtc_base.
BUG=
Review URL: https://codereview.webrtc.org/1789463002
Cr-Commit-Position: refs/heads/master@{#11966}
Permits sending faster when having an attached track before actually
sending since the configured stream is ready to encode as soon as a call
is accepted.
BUG=webrtc:5410
R=deadbeef@webrtc.org
Review URL: https://codereview.webrtc.org/1790703002 .
Cr-Commit-Position: refs/heads/master@{#11963}
