There were a couple of GN and GYP references that were incorrect in Chromium builds:
- GN references between WebRTC targets must be using relative paths, not absolute.
- GYP references between WebRTC targets must be using the <(webrtc_root)v variable
in order to be expanded to the correct path in a Chromium build.
NOTRY=True
TBR=hjon@webrtc.org, hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1681493002
Cr-Commit-Position: refs/heads/master@{#11521}
Reason for revert:
Chromium FYI turns red.
Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> Note: This is a re-land of
> https://codereview.webrtc.org/1660403004/. Reverting it
> was not necessary.
>
> TBR=kjellander@webrtc.org
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/10b9dd7ab1a8c3f80b2d2924be658e43131a4fbe
> Cr-Commit-Position: refs/heads/master@{#11517}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1675113002
Cr-Commit-Position: refs/heads/master@{#11518}
if not on Android/iOS.
This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.
This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).
Note: This is a re-land of
https://codereview.webrtc.org/1660403004/. Reverting it
was not necessary.
TBR=kjellander@webrtc.org
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1674103002
Cr-Commit-Position: refs/heads/master@{#11517}
The files that are built when use_gtk==1 are not included in the Chromium build
(webrtc/media/devices/gtkvideorenderer.cc and webrtc/media/devices/gtkvideorenderer.h)
so to preserve previous behavior in Chromium before/after
https://codereview.webrtc.org/1587193006, this is the right thing to do.
The reason this was discovered was that a Chrome OS build started failing, since
it was lacking the gtk+2.0 package.
NOTRY=True
BUG=chromium:584722
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1677693002
Cr-Commit-Position: refs/heads/master@{#11510}
Reason for revert:
Trybots red? Don't have time to intvestigate
Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/7cd94f66ebfe5bf808d7dcb8da069d35f4a2b31a
> Cr-Commit-Position: refs/heads/master@{#11506}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1677623002
Cr-Commit-Position: refs/heads/master@{#11508}
if not on Android/iOS.
This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.
This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1660403004
Cr-Commit-Position: refs/heads/master@{#11506}
Removes scoped_ptrs and resets, preventing some heap allocation but also
overall showing that these objects won't be reconstructed on the fly.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1670123002 .
Cr-Commit-Position: refs/heads/master@{#11503}
Added accessor and Parse function
removed dependencies on structures from rtcp_utility.h (except RtcpCommonHeader)
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1675583002
Cr-Commit-Position: refs/heads/master@{#11502}
Extracts shared members outside the two objects, removing PayloadRouter
from receivers and the VCM for ViEChannel from senders.
Removes Start/StopThreadsAndSetSharedMembers that was used to set the
shared state between them.
Also adding DCHECKs to document what's only used by the
sender/receiver side.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1654913002 .
Cr-Commit-Position: refs/heads/master@{#11500}
Permits measuring encoding time even when performed on another thread,
typically for hardware encoding, instead of assuming that encoding is
blocking the calling thread.
Permitted encoding time is increased for hardware encoders since they
can be timed to keep 30fps, for instance, without indicating overload.
Merges EncodingTimeObserver into EncodedFrameObserver to have one post-encode
callback.
BUG=webrtc:5042, webrtc:5132
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1569853002 .
Cr-Commit-Position: refs/heads/master@{#11499}
This CL fixes a race where for Thread objects the parent MessageQueue
constructor registers the object in the MessageQueueManager even though
the Thread is not constructed completely yet. Same happens during
destruction.
BUG=webrtc:1225
Review URL: https://codereview.webrtc.org/1666863002
Cr-Commit-Position: refs/heads/master@{#11497}
This is needed because the target is defined in webrtc/common.gyp
and its current location crosses package boundaries when generating
projects for some build systems.
NOTRY=True
Review URL: https://codereview.webrtc.org/1665603003
Cr-Commit-Position: refs/heads/master@{#11496}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
std::vector<rtcp::TmmbItem> will replace TMMBRSet class for storage, processing and preparing TMBBR/TMMBN
(i.e. this TmmbItem replaces Timber structure introduced in https://codereview.webrtc.org/1474693002 )
Previous structures store bitrate in kbps. TmmbItem use bps removing need to regularly divide and multiply by 1000.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1576223003
Cr-Commit-Position: refs/heads/master@{#11491}
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.
TBR=mflodman@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1604563002
Cr-Commit-Position: refs/heads/master@{#11487}
active_layer_ could be dereferenced while being -1...
Also added som DCHECKs
BUG=webrtc:5490
Review URL: https://codereview.webrtc.org/1656233002
Cr-Commit-Position: refs/heads/master@{#11486}
Sparse macro is replaced and new implementation in metrics.h is used.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1564923008
Cr-Commit-Position: refs/heads/master@{#11483}
This CL fixes an issue where the "writable" flag didn't stay set after
::send or ::sendto only sent a partial buffer.
Also SocketTest::TcpInternal has been updated to use rtc::Buffer instead
of manually allocating data.
BUG=webrtc:4898
Review URL: https://codereview.webrtc.org/1616153007
Cr-Commit-Position: refs/heads/master@{#11480}
There is a use case with external codec factories that only support
encoding but not decoding for a given type. This leads to a crash
due to null being registered as codec (after a DCHECK).
This CL adds a NullVideoDecoder that is used instead of the null to
not crash but log to LS_ERROR.
BUG=webrtc:5249
Review URL: https://codereview.webrtc.org/1657023002
Cr-Commit-Position: refs/heads/master@{#11475}
Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name.
The idea is to be able to turn off H264 from chromium with this function because...
1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled.
2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable.
3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not.
The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory.
BUG=chromium:500605, chromium:468365
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1657273002
Cr-Commit-Position: refs/heads/master@{#11474}
Adds negotiation of rtx codecs for red and vp9. To keep backwards
compatibility with older Chrome versions, this change includes two
hacks:
1. Red packets will be retransmitted over the rtx codec associated with
vp8 if no rtx codec is associated with red. This is how Chrome does
it today and ensures that we still can send red over rtx to older
versions.
2. If rtx packets associated with the media codec (vp8/vp9 etc) are
received and red has been negotiated, we will assume that the sender
incorrectly has packetized red inside the rtx header associated with
media. We will therefore restore it with the red payload type
instead, which ensures that we can still receive rtx associated with
red from old versions.
Offering multiple rtx codecs to older versions should not be a problem
since old versions themselves only try to negotiate rtx for vp8.
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:4024
TEST=Verified by running apprtc and emulating packet loss between Chrome with and without the patch.
Review URL: https://codereview.webrtc.org/1649493004 .
Cr-Commit-Position: refs/heads/master@{#11472}
It's generated by some encoders between SPS/PPS and an IDR frame, so we should treat it like sps/pps.
BUG=
Review URL: https://codereview.webrtc.org/1664733002
Cr-Commit-Position: refs/heads/master@{#11470}
* SSRCDatabase doesn't need to be a global instance, so I've changed it to be a "regular" class (i.e. construct via ctor, not maybe via GetSSRCDatabase( + release via ReturnSSRCDatabase())). If we ever have parallel tests running in the same process, they won't have the problem of using the same ssrc database.
* Made RtpSender a more const. Also added some todos for myself and holmer to look into clarifying the threading model.
* Switched from CriticalSectionWrapper to rtc::CriticalSection
* Changed the random seeding to use TickTime::Now().Ticks() since TimeInMicroseconds() could return 0 when the process was starting. This is what TimeInMicroseconds() does anyway but now we don't need to access a global clock object.
BUG=webrtc:3062
Review URL: https://codereview.webrtc.org/1623543002
Cr-Commit-Position: refs/heads/master@{#11462}
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.
In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.
BUG=chromium:500605, chromium:468365, webrtc:5427
Review URL: https://codereview.webrtc.org/1639273002
Cr-Commit-Position: refs/heads/master@{#11456}
For example, when the TURN port has an ALLOCATE_MISMATCH error.
BUG=webrtc:5432
Review URL: https://codereview.webrtc.org/1595613004
Cr-Commit-Position: refs/heads/master@{#11453}
Some WebRTC client had a problem with the change "Removing webrtc::AudioFrame::energy_". Now it is solved.
This reverts commit 2bdcfadc8abd418a30dd5cdf54ba45a429d3d9bf.
BUG=webrtc:3315
Review URL: https://codereview.webrtc.org/1638553003
Cr-Commit-Position: refs/heads/master@{#11448}
A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1652983002
Cr-Commit-Position: refs/heads/master@{#11447}
Reason for revert:
May be the reason for mac_asan timeout
Original issue's description:
> Changed test to validate rtp timstamps not just in RTP packets but also in RTCP Sender Reports.
> Altered it to accept negative value since it is normal for RTCP packet coming before RTP packet to have slightly later time.
>
> BUG=webrtc:5433
>
> Committed: https://crrev.com/f4b9c775122b463db7eb2c4101603759a0d00caf
> Cr-Commit-Position: refs/heads/master@{#11417}
TBR=pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5433
Review URL: https://codereview.webrtc.org/1652973002
Cr-Commit-Position: refs/heads/master@{#11446}
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.
Also moves sending transport feedback to the pacer thread.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1628683002
Cr-Commit-Position: refs/heads/master@{#11443}
We have seen an instance of flakiness of the perf tests where it looked
like timestamp wraparound could be an issue. Better safe...
BUG=
Review URL: https://codereview.webrtc.org/1645463002
Cr-Commit-Position: refs/heads/master@{#11440}
