Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
Prevents double-initialization of decoders due to resolution changes
between initial database settings and first incoming frame.
BUG=webrtc:5251
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1474193002 .
Cr-Commit-Position: refs/heads/master@{#10822}
Also adds a RTC_CHECK in VideoReceiveStream that verifies that decoders
aren't null, since this will attempt to deregister a codec which would
previously fail with an obscure stack trace not indicating what actually
was wrong.
BUG=webrtc:5249
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1479793002 .
Cr-Commit-Position: refs/heads/master@{#10821}
Mostly moved code around in WebRtcVoiceEngine:
- Added new internal class WebRtcVoiceCodecs for static codec functions and the CodecPrefs.
- ConstructCodecs() -> WebRtcVoiceCodecs::SupportedCodecs().
- FindWebRtcCodec -> WebRtcVoiceCodecs::ToCodecInst().
- WebRtcVoiceMediaChannel::SetRecvCodecsInternal() folded into WebRtcVoiceMediaChannel::SetRecvCodecs() (slight logic change).
- Change to how SetRecPayloadType() is implemented in fakewebrtcvoiceengine.h (lines 460-470).
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1461333002
Cr-Commit-Position: refs/heads/master@{#10819}
Also changes presubmit script to not run cpplint on objc dirs.
BUG=
Review URL: https://codereview.webrtc.org/1467173006
Cr-Commit-Position: refs/heads/master@{#10815}
Seconds and fractions parts of the ntp time presented with two values, but used as one.
This helper structure can make that use more clear.
(initially introduced into rtp_rtcp as https://codereview.webrtc.org/1435833003)
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1482593002
Cr-Commit-Position: refs/heads/master@{#10814}
The webrtc/ rule required a ^ to avoid matching e.g. talk/app/webrtc.
Also modify my subscriptions a bit.
R=kjellander@webrtc.org
TEST=verified with depot_tools/watchlists.py
NOTRY=true
Review URL: https://codereview.webrtc.org/1473983002
Cr-Commit-Position: refs/heads/master@{#10813}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Enable cpplint and have it use a whitelist that also checks
in subdirectories.
Move the cpplint check so it runs before the pylint check
since that one always run and increases the time to errors
for cpplint.
Fix all cpplint errors in webrtc/video_engine.
BUG=webrtc:5149
TESTED=Fixed issues reported by:
find webrtc/video_engine -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.
R=pbos@chromium.org, phoglund@chromium.orgTBR=pbos@webrtc.org, phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1481723003 .
Cr-Commit-Position: refs/heads/master@{#10808}
Insted of using a fixed frame rate, we allow the camera to use a lower frame rate. The camera will choose depending on lightning condition.
TESTED= In a room with low light on N5, N6 N7, Galaxy 4.
BUG=webrtc:5262
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1479563004 .
Cr-Commit-Position: refs/heads/master@{#10807}
This cl make it possible for the hw video encoder to downscale a texture image before encoding. The purpose is to allow downscaling if the quality is too bad at the current resolution.
BUG=webrtc:4993
R=magjed@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1470043002 .
Cr-Commit-Position: refs/heads/master@{#10804}
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
This is used on Mac to link against a precompiled libc++
library on Mac (see https://crbug.com/400091 for details)
TBR=tkchin@webrtc.org
TESTED=Verified the warnings like:
ld: warning: directory not found for option '-L../../third_party/libc++-static'
are no longer printed on Mac.
Review URL: https://codereview.webrtc.org/1475643010 .
Cr-Commit-Position: refs/heads/master@{#10797}
Due to Chromium moving over to building with a sysroot
image on Linux in
a931efd5dc
we need to disable that until http://crbug.com/561584 is fixed
(libudev.h is missing and is used by talk/media/devices/libudevsymboltable.h).
Change log: 68cf0b8..aa8e58a
Full diff: 68cf0b8..aa8e58a
No dependencies changed.
No update to Clang.
BUG=chromium:561584
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1468313006
Cr-Commit-Position: refs/heads/master@{#10795}
Removes log disabling under Chromium which doesn't compile due to
missing LS_INFO in the override log implementation.
Also removes dependency on webrtc/test/BUILD.gn which doesn't build in
Chromium (due to third_party/gflags not being present). Instead the
no-op implementation of field_trials in system_wrappers is used.
BUG=chromium:561667, webrtc:4771
R=kjellander@webrtc.orgTBR=henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1473713004 .
Cr-Commit-Position: refs/heads/master@{#10793}
The corresponding set of overrides weren't moved when logging.cc etc.
was moved over. This wasn't noticed because all existing targets before
webrtc fuzzers used to link both rtc_base and rtc_base_approved in
Chromium. Also adding //base:base as a dependency, this used to be
linked in by other targets either way before but generated build errors
when a target solely depends on rtc_base_approved.
BUG=webrtc:4771
R=kjellander@webrtc.orgTBR=henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1473223005 .
Cr-Commit-Position: refs/heads/master@{#10792}
Changes after "git merge cb3f9bd"
* git mv old Android.mk from src/ to webrtc/
* Remove old unused files in src/*.
* Modify webrtc/.gitignore to keep *.mk files.
* Copy old files from master, lost in auto-merge.
src/modules/audio_processing/test/unit_test.cc
src/modules/audio_coding/codecs/isac/fix/test/{Android.mk,kenny.c}
to webrtc, but most of the old test code do not compile with new
webrtc API and are commented out.
* Move src/modules/audio_processing/test/android/apmtest/jni/*.mk to
webrtc/... but the Android.mk files does not work.
Commented out its build target.
* Changes to Android.mk files:
* Change references of src/ to webrtc/.
* Fix include path
* Fix source file list, remove old non-existing files,
add new source files to resolve link errors.
* Add new Android.mk files to build some new static libraries
to link into current Android webrtc .so files.
* Remove unnecessary LOCAL_SHARED_LIBRARIES in Android.mk files
that build static libraries.
* Remove old unnecessary clang workarounds like
-Wno-tautological-pointer-compare
-no-integrated-as
* Fix include path of debug.pb.h in some source files.
* Add -DWEBRTC_POSIX in android-webrtc.mk
* Manually merge Android specific changes in
src/typedefs.h to webrtc/typedefs.h
* Fix trivial syntax error in scoped_ptr.h, calling static_assert.
* Use -std=c++0x in webrtc/system_wrappers/source/Android.mk
* #undef getchaar in spreadsort.hpp
* Verified and not to carry old Android hacks from src/... to webrtc/...
src/system_wrappers/source/android/cpu-features.c
src/modules/interface/module.h
src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
src/system_wrappers/source/trace_posix.cc
src/typedefs.h
More pathes from Alex Luebs:
* Use new unit test kenny.cc.
Delete old kenny.cc.
Comment out unessential code in kenny.cc to fix link error for now.
* Replace old unit test files with new ones in
webrtc/modules/audio_processing/Android.mk.
Delete old audio_processing/test/unit_test.cc.
* Fix compilation errors in
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
Change-Id: I7bbf776eeb9dcfa21a82dd1f2dec378235cbbc3e
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
This means that if a TURN server denies permission for an
unreachable address, we'll no longer ping it fruitlessly.
BUG=webrtc:4917
Review URL: https://codereview.webrtc.org/1415313004
Cr-Commit-Position: refs/heads/master@{#10789}
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
This overrides the default (10.6) in Chromium's
build/common.gypi. It's needed since we want ARC and libc++.
TESTED=Ran webrtc/build/gyp_webrtc before this patch and then
grep -r macosx-version-min out/Debug/* | grep 10.6
which gave a lot of output.
Then with this patch applied, there were no output for 10.6 (only 10.7).
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1474623002 .
Cr-Commit-Position: refs/heads/master@{#10783}
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
