This CL should not change the behaviour of the decoder. The purpose is to prepare for lifetime management of textures received from the SurfaceTexture. The main change is to only use exceptions for error signaling in MediaCodecVideoDecoder.dequeueOutputBuffer() and MediaCodecVideoDecoder.releaseOutputBuffer(), not both exceptions and error return values.
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1383983003 .
Cr-Commit-Position: refs/heads/master@{#10148}
Always send the CL to the CQ, except if --skip-cq is provided.
Add extra CQ trybots, since the baremetal ones are no longer the default.
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1375153004 .
Cr-Commit-Position: refs/heads/master@{#10146}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
In order to minimize bisection ranges when the roll breaks, we want
to update the chromium_revision as often as possible.
Even if a roll doesn't bring in changed dependencies or Clang version,
it still brings in changes in the Chromium build toolchain.
The description now contains links to all dependencies that change
(if any).
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1381963003 .
Cr-Commit-Position: refs/heads/master@{#10141}
Add a TBR= field to enable automated rolling. In some cases, add a
committer to the field, for other rolls: leave it empty.
Always run 'git pull', even if --dry-run is specified, as
it is often used to just generate an updated commit message to
update an existing CL with when updating it to match fixes in Chromium.
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1369333010 .
Cr-Commit-Position: refs/heads/master@{#10140}
Since LKGR is sometimes lagging behind, always use HEAD to
minimize the number of commits to bisect when something breaks for us.
As long as the rolls are passing our CQ, we should be fine.
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1379173002 .
Cr-Commit-Position: refs/heads/master@{#10139}
This CL changes the threshold where we consider a block to be static and
of sufficient quality to not spend bits/CPU encoding it.
Perf note: This change may result in a minor degradation of PSNR/SSIM
and available send bitrate. CPU usage and bitrate sent should however
be greately reduced.
BUG=webrtc:5015
Review URL: https://codereview.webrtc.org/1383533002
Cr-Commit-Position: refs/heads/master@{#10134}
In particular, if 14 short deltas were inserted (2 * capacity of status
vector chunk with 2bit items) followed by a large delta, that status
item would be dropped.
BUG=
Review URL: https://codereview.webrtc.org/1367193002
Cr-Commit-Position: refs/heads/master@{#10132}
This CL is a baby step towards consolidating the timestamps in cricket::VideoFrame and webrtc::VideoFrame, so that we can unify the frame classes in the future.
The elapsed time functionality is not really used. If a video sink wants to know the elapsed time since the first frame they can store the first timestamp themselves and calculate the time delta to later frames. This is already done in all video sinks that need the elapsed time. Having redundant timestamps in the frame classes is confusing and error prone.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1324263004
Cr-Commit-Position: refs/heads/master@{#10131}
The GN changes in
7a73be6eb4
broke libvpx, so a fix in https://codereview.chromium.org/1368223002
was needed to unblock this roll.
Relevant changes:
* src/buildtools: 8d89c1b..0c88009
* src/third_party/boringssl/src: 4c60d35..8c9b878
* src/third_party/libvpx_new/source/libvpx: 90a109f..7d28d12
* src/third_party/libyuv: 62c49dc..d039ad6
* src/third_party/openmax_dl: 2eb98d8..37b900c
Details: 8cf53d6..681f0cd/DEPS
Clang version was not updated in this roll.
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1375233004
Cr-Commit-Position: refs/heads/master@{#10129}
On some devices (confirmed Samsung) the focus mode is not configured correctly by default.
The fix explicitly set the focus mode to FOCUS_MODE_CONTINUOUS_VIDEO if this mode is supported.
BUG=webrtc:4991
Review URL: https://codereview.webrtc.org/1338773002
Cr-Commit-Position: refs/heads/master@{#10128}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.
(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368843003
Cr-Commit-Position: refs/heads/master@{#10127}
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
This change filters out local ports when CF_HOST is not originally specified to prevent these ports from sending out STUN which leaks IP address.
BUG=webrtc:4946
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1378753003 .
Cr-Commit-Position: refs/heads/master@{#10121}
Connecting TransportChannelImpls directly to the TransportController,
and removing redundant signal forwarding/state aggregating code from
Transport. This brings us closer to just getting rid of Transport
entirely.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1380563002 .
Cr-Commit-Position: refs/heads/master@{#10120}
This CL refactors RendererCommon.getSamplingMatrix() so it does not have any dependecy to SurfaceTeture. The purpose is to prepare for a change in how texture frames are represented - only the texture matrix will be exposed, not the SurfaceTexture itself. This CL also adds an extra test for RendererCommon.rotateTextureMatrix().
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1375593002 .
Cr-Commit-Position: refs/heads/master@{#10118}
Reason for revert:
Relanding with SetConfiguration not pure virtual.
Original issue's description:
> Revert of Adding PeerConnectionInterface::SetConfiguration method. (patchset #4 id:60001 of https://codereview.webrtc.org/1317353005/ )
>
> Reason for revert:
> Broke FYI bots because SetConfiguration is pure virtual and MockPeerConnectionImpl doesn't implement it. Need to reland with SetConfiguration not pure virtual.
>
> Original issue's description:
> > Adding PeerConnectionInterface::SetConfiguration method.
> >
> > Also updated the JNI and Objective-C bindings. Later, will have a CL to
> > remove UpdateIce, which this method effectively replaces.
> >
> > BUG=webrtc:4945
> >
> > Committed: https://crrev.com/70702afbcb8418fe93747e7ed63bcbf5e56b90e9
> > Cr-Commit-Position: refs/heads/master@{#10040}
>
> TBR=guoweis@webrtc.org,pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4945
>
> Committed: https://crrev.com/7603c76ab077b1e2033bb179595129bd96797345
> Cr-Commit-Position: refs/heads/master@{#10041}
TBR=guoweis@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4945
Review URL: https://codereview.webrtc.org/1361273002
Cr-Commit-Position: refs/heads/master@{#10112}
The purpose of this CL is to use jlongFromPointer() for converting frame pointers to jlong instead of implicit casts which is not safe.
In order to respect constness, I had to make a small helper function for this.
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1373233002 .
Cr-Commit-Position: refs/heads/master@{#10108}
It turns out that if a bot is interrupted during test execution,
the existence of this directory causes the next build to fail.
This usually happens when the Buildbot master is restarted.
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1376723002 .
Cr-Commit-Position: refs/heads/master@{#10105}
Reason for revert:
The top row in the video stream from the camera is messed up. The byte[] pointer is not the same as GetDirectBufferAddress() apparently.
Original issue's description:
> Android VideoCapturer: Send ByteBuffer instead of byte[]
>
> The purpose with this CL is to replace GetByteArrayElements() and ReleaseByteArrayElements() with GetDirectBufferAddress().
>
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/cb3649b40b3fd6d5bbb0a92003b717e46ce90924
> Cr-Commit-Position: refs/heads/master@{#10091}
TBR=hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1377783002
Cr-Commit-Position: refs/heads/master@{#10103}
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).
Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).
Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/
Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).
Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS
Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh
BUG=481034, 535973
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1355083002
Cr-Commit-Position: refs/heads/master@{#10101}
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.
It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.
Review URL: https://codereview.webrtc.org/1351803002
Cr-Commit-Position: refs/heads/master@{#10100}
