Commit Graph

10180 Commits

Author SHA1 Message Date
7623ce4aeb Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.

Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}

BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1510183002 .

Cr-Commit-Position: refs/heads/master@{#10948}
2015-12-09 11:13:40 +00:00
d3c944755e Nuke TickTime::UseFakeClock.
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.

BUG=webrtc:5318
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1512853002 .

Cr-Commit-Position: refs/heads/master@{#10947}
2015-12-09 10:21:09 +00:00
bda7e0b932 Fixing issue with default stream upon setting 2nd remote description.
If a description is set that requires making a default stream, and one
already exists, we'll now keep the existing default audio/video tracks,
rather than destroying them and recreating them. Destroying them caused
the blink MediaStream to go to an "ended" state, which is the root cause
of the bug.

BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1469833006

Cr-Commit-Position: refs/heads/master@{#10946}
2015-12-09 01:13:53 +00:00
d02b0fab76 Add oldest rotation type option to RTCFileLogger
BUG=

Review URL: https://codereview.webrtc.org/1432753003

Cr-Commit-Position: refs/heads/master@{#10945}
2015-12-08 21:59:11 +00:00
5e465c33ca Make NoiseSuppression not a processing component (bit exact).
BUG=webrtc:5298

patch from issue 1490333004 at patchset 1 (http://crrev.com/1490333004#ps1)

Review URL: https://codereview.webrtc.org/1507683006

Cr-Commit-Position: refs/heads/master@{#10944}
2015-12-08 21:22:35 +00:00
1a9d615cbf Add tracing to public PeerConnection methods.
Adds tracing specifically to Close, for creating streams and also moves
tracing for SetLocal/RemoteDescription from WebRtcSession. Also adding
some tracing in ChannelManager to see what's taking time inside Close.

BUG=webrtc:5167
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1509903002 .

Cr-Commit-Position: refs/heads/master@{#10943}
2015-12-08 21:15:26 +00:00
2d63680d3e Roll chromium_revision 9dfb3a1..dad6346 (363718:363782)
Change log: 9dfb3a1..dad6346
Full diff: 9dfb3a1..dad6346

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1506853005

Cr-Commit-Position: refs/heads/master@{#10942}
2015-12-08 21:06:46 +00:00
7b2f7627e4 Don't call SetPreviewFormat if capturing to textures.
This fix an issue seen on Huawei Y300 where the camera feed is black and white if we capture to textures and setpreviewformat is called.

BUG=webrtc:4993

Review URL: https://codereview.webrtc.org/1502223002

Cr-Commit-Position: refs/heads/master@{#10941}
2015-12-08 20:03:07 +00:00
edd8fefa9b Add new view that renders local video using AVCaptureLayerPreview.
BUG=

Review URL: https://codereview.webrtc.org/1497393002

Cr-Commit-Position: refs/heads/master@{#10940}
2015-12-08 19:08:44 +00:00
70f9903e57 Make HighPassFilter not a ProcessingComponent anymore (bit exact).
BUG=webrtc:5298

Review URL: https://codereview.webrtc.org/1490333004

Cr-Commit-Position: refs/heads/master@{#10939}
2015-12-08 19:07:38 +00:00
246b8171a6 Refactor handling of AudioOptions.
- Remove MediaEngineInterface::GetAudioOptions(), SetAudioOptions() and SetSoundDevices().
- Remove the WebRtcVoiceEngine infrastructure for those calls.

BUG=webrtc:4690
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1500633002

Cr-Commit-Position: refs/heads/master@{#10938}
2015-12-08 17:50:33 +00:00
8237abf563 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.

Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}

TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695

Review URL: https://codereview.webrtc.org/1507903005

Cr-Commit-Position: refs/heads/master@{#10937}
2015-12-08 15:12:11 +00:00
e10c82dc12 Deletes temporary files that are generated in several ACM unittests.
This applies to AcmSwitchingOutputFrequencyOldApi.*,
AcmReceiverBitExactnessOldApi.* and AcmSenderBitExactnessOldApi.*.

BUG=webrtc:4647
NOTRY=true

Review URL: https://codereview.webrtc.org/1503043003

Cr-Commit-Position: refs/heads/master@{#10936}
2015-12-08 13:03:32 +00:00
d7b7ae8bda Add encode/decode time tracing to audio_coding.
Also removes virtual from VideoDecoder::Decode and updated mocks and
tests accordingly to use VideoDecoder::DecodeInternal instead.

BUG=webrtc:5167
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1512483003 .

Cr-Commit-Position: refs/heads/master@{#10935}
2015-12-08 12:41:44 +00:00
9f45a45a62 Add tracing to upper-level WebRTC calls.
Adds tracing to WebRtcSession and corresponding BaseChannel calls to see
where time is spent better.

BUG=webrtc:5167
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1505023003 .

Cr-Commit-Position: refs/heads/master@{#10934}
2015-12-08 12:26:11 +00:00
cd6f539a08 Revert of RTCCertificate::Expires() and ::HasExpired() implemented (patchset #5 id:140001 of https://codereview.webrtc.org/1494103003/ )
Reason for revert:
RTCCertificate's expires_timestamp_ns was renamed to Expires but the old function is still used in one place in Chromium...
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Mac%20Builder/builds/7405

Original issue's description:
> RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime().
>
> NOPRESUBMIT=true
> BUG=chromium:544894
>
> Committed: https://crrev.com/20ef654174e245b3a06c9e9045bb97be9acd90cf
> Cr-Commit-Position: refs/heads/master@{#10930}

TBR=torbjorng@webrtc.org,hta@webrtc.org,kjellander@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:544894

Review URL: https://codereview.webrtc.org/1506883005

Cr-Commit-Position: refs/heads/master@{#10933}
2015-12-08 10:32:19 +00:00
fe32a76d60 Create fuzzer tests for audio decoders
This change adds fuzzer tests for iLBC, iSAC fix and float, and
Opus. The fuzzer function takes a random input vector and splits it
into a number of payloads. The lengths of the payloads is also
determined by the random vector. The payloads are decoded with the
decoders.

BUG=webrtc:5306
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1499093002 .

Cr-Commit-Position: refs/heads/master@{#10932}
2015-12-08 10:27:34 +00:00
ffea13c42c PRESUBMIT: change native API check from warning to information.
NOTRY=True

Review URL: https://codereview.webrtc.org/1507723002

Cr-Commit-Position: refs/heads/master@{#10931}
2015-12-08 09:57:24 +00:00
20ef654174 RTCCertificate::Expires() and ::HasExpired() implemented using SSLCertificate::CertificateExpirationTime().
NOPRESUBMIT=true
BUG=chromium:544894

Review URL: https://codereview.webrtc.org/1494103003

Cr-Commit-Position: refs/heads/master@{#10930}
2015-12-08 09:42:46 +00:00
325b34542d There was an old scaling for CNG 48 kHz in the code, from the time where Audio Coding Module didn't have full 48 kHz support. This CL removes the scaling.
The bug hasn't caused us any problems, since we don't run CNG together with Opus (our only real 48 kHz codec), but would cause problems if used with PCB16b @ 48 kHz.

BUG=webrtc:5303
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1496243002 .

Cr-Commit-Position: refs/heads/master@{#10929}
2015-12-08 09:13:08 +00:00
88eeac4aa7 Adding video_processing to presubmit lint check
BUG=webrtc:5259
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1507213002 .

Cr-Commit-Position: refs/heads/master@{#10928}
2015-12-08 08:21:39 +00:00
4654d204e4 Add test which verifies that the RTP header extensions are set correctly for FEC packets.
Also taking the opportunity to do a little bit of clean up.

BUG=webrtc:705
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1506863002 .

Cr-Commit-Position: refs/heads/master@{#10927}
2015-12-08 08:10:58 +00:00
03ef053202 Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1506773002 .

Cr-Commit-Position: refs/heads/master@{#10926}
2015-12-08 08:09:07 +00:00
99ab9447d1 Clang format of video_processing folder.
BUG=webrtc:5259

Review URL: https://codereview.webrtc.org/1508793002

Cr-Commit-Position: refs/heads/master@{#10925}
2015-12-08 06:54:59 +00:00
a440c6fa80 Roll chromium_revision 3b8be21..9dfb3a1 (363445:363718)
Change log: 3b8be21..9dfb3a1
Full diff: 3b8be21..9dfb3a1

Changed dependencies:
* src/tools/gyp: e2313c0..70ee80e
DEPS diff: 3b8be21..9dfb3a1/DEPS

No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1507643003

Cr-Commit-Position: refs/heads/master@{#10924}
2015-12-08 06:26:33 +00:00
386869247f Free SCTP data channels asynchronously in PeerConnection.
This is needed so that the data channel isn't deleted while one of its
own methods is on the call stack.

BUG=565048

Review URL: https://codereview.webrtc.org/1492383002

Cr-Commit-Position: refs/heads/master@{#10923}
2015-12-07 23:32:31 +00:00
46ad5426b0 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.

.../webrtc/base/atomicops.h:71:8: note:   no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'

Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1505053002

Cr-Commit-Position: refs/heads/master@{#10922}
2015-12-07 22:29:21 +00:00
6f28cf0b95 Implement standalone event tracing in AppRTCDemo.
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.

BUG=webrtc:5158
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1457383002 .

Cr-Commit-Position: refs/heads/master@{#10921}
2015-12-07 22:17:26 +00:00
84f0970d10 Reland of "Create rtc::AtomicInt POD struct."
Relands https://codereview.webrtc.org/1420043008/ with brace initializers
instead of constructors hoping that they won't introduce static
initializers.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1498953002 .

Cr-Commit-Position: refs/heads/master@{#10920}
2015-12-07 22:07:11 +00:00
0f490a5b86 Add logs when stun or turn host lookup is completed.
This will help investigate issues caused by DNS lookup.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1493863002 .

Cr-Commit-Position: refs/heads/master@{#10919}
2015-12-07 20:06:27 +00:00
cd4003f3df Use @webrtc.org addresses for OWNERS.
Fixes talk/app/webrtc/OWNERS and removes houssainy@google.com from
webrtc/tools/rtcbot/OWNERS.

BUG=
R=andresp@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1505613004 .

Cr-Commit-Position: refs/heads/master@{#10918}
2015-12-07 18:53:25 +00:00
cf890bc58e Roll gtest-parallel.
Brings in fixes that save log output to disk instead of piping them
through Python. Should fix problem where output from tests stall for
more than 10 seconds.

Also enabling JsepPeerConnectionP2PTestClient on all platforms again.

BUG=webrtc:5231
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1509463002 .

Cr-Commit-Position: refs/heads/master@{#10917}
2015-12-07 15:45:08 +00:00
0608dc5db5 Roll chromium_revision 4918765..3b8be21 (363393:363445)
Change log: 4918765..3b8be21
Full diff: 4918765..3b8be21

No dependencies changed.
No update to Clang.

TBR=
NOTRY=True

Review URL: https://codereview.webrtc.org/1503973002

Cr-Commit-Position: refs/heads/master@{#10916}
2015-12-07 15:41:31 +00:00
5f6deaf525 Remove unused RTP-header parser.
D'oh.

BUG=
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1506743003 .

Cr-Commit-Position: refs/heads/master@{#10915}
2015-12-07 15:18:18 +00:00
ab82cbbfd9 Disable RtcEventLogTest.DropOldEvents on memcheck.
BUG=webrtc:5312
TBR=terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1506763003 .

Cr-Commit-Position: refs/heads/master@{#10914}
2015-12-07 14:30:22 +00:00
03671cb38a Use existing parser in ReceivesAndRetransmitsNack.
Removes logspam of "Failed to find extension id:".

BUG=
TBR=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1502993003 .

Cr-Commit-Position: refs/heads/master@{#10913}
2015-12-07 14:22:34 +00:00
fc47ed6c05 rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1496883002 .

Cr-Commit-Position: refs/heads/master@{#10912}
2015-12-07 13:46:42 +00:00
1aa420b6aa Remove avg encode time from CpuOveruseMetric struct and use value from OnEncodedFrame instead.
BUG=

Review URL: https://codereview.webrtc.org/1278383002

Cr-Commit-Position: refs/heads/master@{#10911}
2015-12-07 11:12:27 +00:00
9d69c3f4d9 Return a copy of the supported RTP header extensions instead of a reference.
This also renames the method to better reflect what it does.

BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1486123002 .

Cr-Commit-Position: refs/heads/master@{#10910}
2015-12-07 09:45:49 +00:00
b86d4e4a8d Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.

BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1479023002 .

Cr-Commit-Position: refs/heads/master@{#10909}
2015-12-07 09:26:32 +00:00
03f80ebb83 Refactor EglBase configuration.
Delete EglBase.ConfigType, instead pass arrays of attributes, and define
constant arrays for the common cases.

Both in progress NativeToI420 and extending GlRectDrawer to other shapes (with alpha) needs this.

BUG=b/25694445

Review URL: https://codereview.webrtc.org/1498003002

Cr-Commit-Position: refs/heads/master@{#10908}
2015-12-07 09:17:22 +00:00
a8565425bc Initial VideoProcessing refactoring.
This CL is the first in a series of CLs to refactor
VideoProcessing(Module) to follow Google C++ style guide and make the
code more readable.

This CL removed inheritance from Module, renames variables and makes
VideoProcessingImpl::PreprocessFrame return a frame pointer if there
is a frame to send, nullptr otherwise. The affected CLs also passes git
cl lint.

BUG=webrtc:5259

Review URL: https://codereview.webrtc.org/1482913003

Cr-Commit-Position: refs/heads/master@{#10907}
2015-12-07 09:10:01 +00:00
2512f44397 Roll chromium_revision 292ab9f..4918765 (363376:363393)
Change log: 292ab9f..4918765
Full diff: 292ab9f..4918765

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1508563002

Cr-Commit-Position: refs/heads/master@{#10906}
2015-12-07 04:20:11 +00:00
c9f1cb84d8 Roll chromium_revision 72c3265..292ab9f (363365:363376)
Change log: 72c3265..292ab9f
Full diff: 72c3265..292ab9f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1508503002

Cr-Commit-Position: refs/heads/master@{#10905}
2015-12-06 20:23:29 +00:00
34a7054fd6 Roll chromium_revision 626eecf..72c3265 (363027:363365)
0c2a94ab87
needed the addition of tools/telemetry and third_party/catapult symlinks
for Android.
GN changes required modification of .gn.

Change log: 626eecf..72c3265
Full diff: 626eecf..72c3265

Changed dependencies:
* src/buildtools: 818123d..b747a9e
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/fde89b4..6d9e5a7
DEPS diff: 626eecf..72c3265/DEPS

No update to Clang.

TBR=
NOTRY=True

Review URL: https://codereview.webrtc.org/1503883002

Cr-Commit-Position: refs/heads/master@{#10904}
2015-12-06 18:32:42 +00:00
1218d7ad2f Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.

TBR=pthatcher@webrtc.org
BUG=webrtc:3618

This is a reland of https://codereview.webrtc.org/1453523002

Review URL: https://codereview.webrtc.org/1505573002 .

Cr-Commit-Position: refs/heads/master@{#10903}
2015-12-05 18:00:04 +00:00
86aaa4be8d Revert "Allow remote fingerprint update during a call"
This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63.

This commit somehow is different from what I have in my local copy. Revert and will recommit.

TBR=pthatcher@webrtc.org
BUG=3618

Review URL: https://codereview.webrtc.org/1494373004 .

Cr-Commit-Position: refs/heads/master@{#10902}
2015-12-05 17:55:54 +00:00
9c38c2d33f Allow remote fingerprint update during a call
Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.

BUG=webrtc:3618
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1453523002 .

Cr-Commit-Position: refs/heads/master@{#10901}
2015-12-05 17:46:16 +00:00
381b4217cb Ping backup connection at a slower rate
and make it configurable from the app.
Changed the decision on whether a connection is pingable:
1.Check whether a connection is a backup connection. A connection is considered as a backup connection only if the channel is complete, the connection is active and it is not the best connection.
2. Ping a non-backup connection if it is active and for backup connection, ping it at a slower rate.
Note the default behavior is the same as before.

Also cached the channel state since we are accessing it more often.
BUG=webrtc:5034
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1455033004 .

Cr-Commit-Position: refs/heads/master@{#10900}
2015-12-04 20:24:10 +00:00
45b0efd378 Stop sending stun binding requests after certain amount of time.
Also stop it if the request timed out.

It is going to be complicated to keep this and make it sync with the connection bind request as they may be on two different ports.

BUG=

Review URL: https://codereview.webrtc.org/1465843004

Cr-Commit-Position: refs/heads/master@{#10899}
2015-12-04 16:57:31 +00:00