Commit Graph

8888 Commits

Author SHA1 Message Date
18f7c8d4df Remove warning suppression from VideoToolboxEncoder
This warning no longer needs to be supressed as the issue
has been fixed.

BUG=webrtc:6307
TBR=tkchin@webrtc.org

Review-Url: https://codereview.webrtc.org/2537293003
Cr-Commit-Position: refs/heads/master@{#15331}
2016-11-30 13:31:07 +00:00
076c0118c5 Change unit of logged bitrate stats in bytes/s to bits/s.
Multiplier added to ToString method in AggregatedStats.

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2535323003
Cr-Commit-Position: refs/heads/master@{#15330}
2016-11-30 13:17:21 +00:00
aff96361d3 Greatly reduce number of level controller tests.
These tests have been causing a large number of false alerts, which is
tons of work for the perf sheriff. It's infeasible to test every single
permutation, so we must choose carefully. This CL reduces the number
of RESULT lines from the test from ~450 to ~50. I attempted to choose
interesting permutations, but you probably know better what's
interesting...

https://chromeperf.appspot.com/report?
sid=a7193c96f708018848ca07ad9c78ac657cadab3c70b3939c42bd7d70a092d61a
also suggests most of the metrics have enormous standard deviations,
so maybe you could look into how stable the metrics really are
and remove/stabilize the ones that aren't?

BUG=chromium:666725

Review-Url: https://codereview.webrtc.org/2529393006
Cr-Commit-Position: refs/heads/master@{#15329}
2016-11-30 13:04:47 +00:00
6a2e20ad14 Make sure GetLastError on a PlatformThread return an error that is relevant to the thread.
BUG=none

Review-Url: https://codereview.webrtc.org/2541003002
Cr-Commit-Position: refs/heads/master@{#15328}
2016-11-30 12:53:14 +00:00
b49fc142e3 RtpDataEngine, FindCodecByName: Don't reassign codecs
BUG=None

Review-Url: https://codereview.webrtc.org/2541583003
Cr-Commit-Position: refs/heads/master@{#15327}
2016-11-30 12:52:10 +00:00
998df1d1a0 Create webrtc/sdk/android folder
BUG=webrtc:5882
NOTRY=True

Review-Url: https://codereview.webrtc.org/2532393004
Cr-Commit-Position: refs/heads/master@{#15326}
2016-11-30 12:51:05 +00:00
78b4d56535 Relanding "Pass time constant to bwe smoothing filter."
An earlier attempt to land this was in https://codereview.webrtc.org/2518923003/

It was failed because it removed an API. This CL fixes this.

BUG=webrtc:6443, webrtc:6303

Review-Url: https://codereview.webrtc.org/2536753002
Cr-Commit-Position: refs/heads/master@{#15325}
2016-11-30 12:47:47 +00:00
8d66a5a3b1 Disabled flaky P2PTestConductor tests on ASAN and MSAN.
TBR=deadbeef@webrtc.org
BUG=webrtc:6776

Review-Url: https://codereview.webrtc.org/2539103002
Cr-Commit-Position: refs/heads/master@{#15324}
2016-11-30 12:23:39 +00:00
0245da0fa0 Move ownership of PacketRouter from CongestionController to Call.
And delete the method CongestionController::packet_router.

BUG=None

Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
2016-11-30 11:35:28 +00:00
06251f1955 Reduce ProbeController::kDefaultMaxProbingBitrateBps to 10 mbps.
BUG=none

Review-Url: https://codereview.webrtc.org/2535903003
Cr-Commit-Position: refs/heads/master@{#15322}
2016-11-30 10:48:46 +00:00
f4a5942e62 Disabled all ScreenCapturerIntegrationTests on Windows
TBR=zijiehe@chromium.org
BUG=webrtc:6666

Review-Url: https://codereview.webrtc.org/2538093002
Cr-Commit-Position: refs/heads/master@{#15321}
2016-11-30 10:39:00 +00:00
3a864d2545 MB: Add swarming bots in the FYI waterfall and remove memcheck swarming.
BUG=chromium:657725
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2532943004
Cr-Commit-Position: refs/heads/master@{#15320}
2016-11-30 10:35:47 +00:00
706a45e68e Added missing include to fix waterfall compile error.
Bots failue caused by https://codereview.webrtc.org/2517243005/

NOTRY=True
TBR=stefan@webrtc.org
BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2544473002
Cr-Commit-Position: refs/heads/master@{#15318}
2016-11-30 09:53:19 +00:00
f15a2c5193 Delete deprecated versions of Copy, ScaleFrom and CropAndScaleFrom.
BUG=webrtc:6672

Review-Url: https://codereview.webrtc.org/2528143002
Cr-Commit-Position: refs/heads/master@{#15317}
2016-11-30 09:52:12 +00:00
0583b286e4 Collecting RTCIceCandidatePairStats.transport_id and improved unittests.
RTCIceCandidatePairStats.transport_id is set to the related
RTCTransportStats' id.

Unittest for RTCIceCandidatePairStats is updated to do EXPECT_EQ
between actual and an expected hardcoded dictionary. The previous way of
testing, ExpectReportContainsCandidatePair, is removed.

(ExpectReportContainsCandidate still exist, we might want to replace
this by EXPECT_EQ testing in a follow up.)

Unittest for RTCTransportStats is similarly updated and
ExpectReportContainsTransportStats is removed. A bug was uncovered where
the "rtcp_connection_info.best_connection = true" case was not tested
(a copy of rtcp_connection_info was used in the test, modifying that had
no affect on the test) - fixed.

rtcstats_integrationtest.cc updated to take transport_id into account.

In order to reuse an updated version of expected_rt[c]p_transport in the
unittest, timestamps are ignored by RTCStats::operator==.

BUG=chromium:627816, chromium:653873, chromium:653873, webrtc:6755

Review-Url: https://codereview.webrtc.org/2527113002
Cr-Commit-Position: refs/heads/master@{#15316}
2016-11-30 09:50:36 +00:00
0c43f779f8 Update video histograms that do not have a minimum lifetime limit before being recorded.
Updated histograms:
"WebRTC.Video.ReceivedPacketsLostInPercent" (two RTCP RR previously needed)
"WebRTC.Video.ReceivedFecPacketsInPercent" (one received packet previously needed)
"WebRTC.Video.RecoveredMediaPacketsInPercentOfFec" (one received FEC packet previously needed)

Prevents logging stats if call was shortly in use.

BUG=b/32659204

Review-Url: https://codereview.webrtc.org/2536653002
Cr-Commit-Position: refs/heads/master@{#15315}
2016-11-30 09:42:32 +00:00
759e0b7241 Fix memory leak in video_coding::PacketBuffer::InsertPacket.
BUG=webrtc:6788

Review-Url: https://codereview.webrtc.org/2535203002
Cr-Commit-Position: refs/heads/master@{#15314}
2016-11-30 09:32:11 +00:00
be74270ebe Calculate JitterBufferDelayInMs in the new jitter buffer.
JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
2016-11-30 09:31:45 +00:00
e69b46863a Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
Reason for revert:
internal bot failure

Original issue's description:
> Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/84e56d576806635c966093d5421c5d04c9b90746
> Cr-Commit-Position: refs/heads/master@{#15310}

TBR=kwiberg@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2537243004
Cr-Commit-Position: refs/heads/master@{#15312}
2016-11-30 09:19:06 +00:00
1731c9cb4c Use swap instead of copy in RtcHistogram::GetAndReset.
BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2268323002
Cr-Commit-Position: refs/heads/master@{#15311}
2016-11-30 08:29:17 +00:00
84e56d5768 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2411613002
Cr-Commit-Position: refs/heads/master@{#15310}
2016-11-30 08:28:07 +00:00
097529f34c Remove 3 defines in voice_engine_configurations.h
WEBRTC_VOICE_ENGINE_AGC, WEBRTC_VOICE_ENGINE_ECHO, and
WEBRTC_VOICE_ENGINE_NR are now gone.

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2530373002
Cr-Commit-Position: refs/heads/master@{#15309}
2016-11-30 08:12:57 +00:00
e61fbfffda Use RotateDesktopFrame in DirectX capturer
To support rotation in DirectX capturer, several other changes are also
required.
1. Removing AddRect in RotateDesktopFrame, this is a performance improvement.
DxgiOutputDuplicator creates a rotated DesktopRegion, which can be directly
add to updated_region.
2. DxgiOutputDuplicator::SourceRect() is not accurate, the rectangle in source
is controlled by |offset| or |rotation_| + |offset|, instead of desktop_rect().
3. The |region| in DxgiTexture::CopyFrom() is not accurate. It needs an
unrotated DesktopRegion which offsets by |offset| instead of desktop_rect(). To
avoid generating both rotated and unrotated updated_region, this parameter has
been removed. This impacts DxgiTextureStagning performance a little bit (1.5ms).
Refer to bug for details.

BUG=webrtc:6646

Review-Url: https://codereview.webrtc.org/2530303002
Cr-Commit-Position: refs/heads/master@{#15308}
2016-11-30 00:09:57 +00:00
166e59a70f Enable ScreenCapturerIntegrationTests
This change enables ScreenCapturerIntegrationTests.

BUG=webrtc:6666

Review-Url: https://codereview.webrtc.org/2513213002
Cr-Commit-Position: refs/heads/master@{#15307}
2016-11-29 22:46:56 +00:00
c9e80eee51 Adding packet overhead to audio network adaptor.
BUG=webrtc:6303, webrtc:6762

Review-Url: https://codereview.webrtc.org/2530653003
Cr-Commit-Position: refs/heads/master@{#15305}
2016-11-29 21:00:37 +00:00
a33287761a Remove overhead from video bitrate.
BUG=webrtc:6638

Review-Url: https://codereview.webrtc.org/2517243005
Cr-Commit-Position: refs/heads/master@{#15303}
2016-11-29 17:25:10 +00:00
290d43aa14 Add a new UMA metric in APM to track incoming capture-side audio level
This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and
WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once
every 10 seconds.

BUG=webrtc:6622

Review-Url: https://codereview.webrtc.org/2534473004
Cr-Commit-Position: refs/heads/master@{#15300}
2016-11-29 16:09:17 +00:00
939e08f5f4 Added webrtc/audio/utility directory and empty GN target.
This Cl is the first step of a 3-step process to trick the build system of a
webrtc dependency into not failing.

The second step will be to register this folder in the dependency
build system. It cannot be done first, because there must be a GN
dependency from some existing target to a target in the new folder.

The third step is to land https://codereview.webrtc.org/2424173003, in
which the webrtc/audio/utility is used.

When a new folder with a publicly visible header is added, the build
files of said build system must be manually updated to register the
new folder.

BUG=webrtc:6548

Review-Url: https://codereview.webrtc.org/2532403003
Cr-Commit-Position: refs/heads/master@{#15299}
2016-11-29 15:32:15 +00:00
ee414d90b0 Added sanity check to VCMDecodingState::UsingFlexibleMode to prevent OOB error.
BUG=chromium:667504

Review-Url: https://codereview.webrtc.org/2534883003
Cr-Commit-Position: refs/heads/master@{#15298}
2016-11-29 15:01:29 +00:00
768d6259dc Fix spelling mistake in RTP module declaration.
BUG=None
R=danilchap@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2531223004
Cr-Commit-Position: refs/heads/master@{#15296}
2016-11-29 14:19:49 +00:00
b890c95c33 Replace some asserts with DCHECKs
NOPRESUBMIT=true
BUG=webrtc:6779

Review-Url: https://codereview.webrtc.org/2535643002
Cr-Commit-Position: refs/heads/master@{#15295}
2016-11-29 13:30:47 +00:00
5049942219 Refactor RMSLevel and give it new functionality
This change rewrites RMSLevel, making it accept an ArrayView as input,
and modify the implementation somewhat. It also makes the class keep
track of the peak RMS in addition to the average RMS over the
measurement period.

New tests are added to cover the new functionality.

BUG=webrtc:6622

Review-Url: https://codereview.webrtc.org/2535523002
Cr-Commit-Position: refs/heads/master@{#15294}
2016-11-29 12:26:31 +00:00
f17cae24f4 Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
Remove entries for headers that no longer exists in the webrtc/ dir.

BUG=webrtc:5878
NOTRY=True

Review-Url: https://codereview.webrtc.org/2540633002
Cr-Commit-Position: refs/heads/master@{#15292}
2016-11-29 10:52:22 +00:00
668eb3b71c Add overhead to transport feedback observer.
BUG=webrtc:6762

Review-Url: https://codereview.webrtc.org/2525283002
Cr-Commit-Position: refs/heads/master@{#15291}
2016-11-29 10:24:23 +00:00
455b512333 Landmine to clobber Windows builders
Speculative clobber to see if the Win 64-bit Debug linking
errors go away.

BUG=chromium:668961
TBR=ossu@webrtc.org

Review URL: https://codereview.webrtc.org/2538593003 .

Cr-Commit-Position: refs/heads/master@{#15289}
2016-11-29 10:14:47 +00:00
1b5b22dc17 Fixed bug in ExtractFrameFromY4mFile API which was not extracting the frames correctly.
Issue: This API was calculating the file_header and frame_header offset only for the first frame which is not the right logic. We need to skip the file and frame header every time we extract a new frame.

Also added a unit test case which compares the extracted frame with the frame stored in text file.

NOPRESUBMIT=true
NOTRY=true

BUG=webrtc:6761

Review-Url: https://codereview.webrtc.org/2532963002
Cr-Commit-Position: refs/heads/master@{#15288}
2016-11-29 10:01:26 +00:00
db346a7cbe RTCStatsIntegrationTest added.
This is an integration test using peerconnectiontestwrapper.h to set up
and end to end test using a real PeerConnection implementation. These
tests will complement rtcstatscollector_unittest.cc which collects all
stats using mocks.

The integration test is set up so that all stats types are returned by
GetStats and verifies that expected dictionary members are defined. The
test could in the future be updated to include sanity checks for the
values of members. There is a sanity check that references to other
stats dictionaries yield existing stats of the appropriate type, but
other than that members are only tested for if they are defined not.

StatsCallback of rtcstatscollector_unittest.cc is moved so that it can
be reused and renamed to RTCStatsObtainer.

TODO: Audio stream track stats members are missing in the test. Find out
if this is because of a real problem or because of testing without real
devices. Do this before closing crbug.com/627816.

BUG=chromium:627816

Review-Url: https://codereview.webrtc.org/2521663002
Cr-Commit-Position: refs/heads/master@{#15287}
2016-11-29 09:57:08 +00:00
876222f77d Move usage of QualityScaler to ViEEncoder.
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
  encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
  having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
  but has a callback to ViEEncoder that it uses to express it's desire
  for lower resolution.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
2016-11-29 09:44:22 +00:00
320e45ad87 Use RateCounter for input/sent fps stats. Reports average of periodically computed stats over a call.
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
2016-11-29 09:40:46 +00:00
6cf94a0118 Only use BoringSSL time callback in unit tests.
The actual time used in production code should honor the epoch time.

BUG=webrtc:6737

Review-Url: https://codereview.webrtc.org/2526433002
Cr-Commit-Position: refs/heads/master@{#15282}
2016-11-29 01:38:39 +00:00
352444fcac RTC_[D]CHECK_op: Remove superfluous casts
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.

It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
2016-11-28 23:59:03 +00:00
af476c737f RTC_[D]CHECK_op: Remove "u" suffix on integer constants
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
2016-11-28 23:21:51 +00:00
80ed35e21c Implement periodic bandwidth probing in application-limited region.
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.

BUG=webrtc:6332

Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
2016-11-28 21:11:24 +00:00
fd87f4af66 Opus: Move complexity variable out of conditional build flag
BUG=webrtc:6708

Review-Url: https://codereview.webrtc.org/2535933002
Cr-Commit-Position: refs/heads/master@{#15277}
2016-11-28 19:16:00 +00:00
1bc3146e08 Disable more VideoProcessorIntegrationTest tests on Linux 32-bit
The previously disabled tests on TSan and UBSan are also failing on
Linux 32-bit.

BUG=webrtc:6781
R=ehmaldonado@webrtc.org
TBR=deadbeef@webrtc.org

Review URL: https://codereview.webrtc.org/2535913002 .

Cr-Commit-Position: refs/heads/master@{#15276}
2016-11-28 18:34:43 +00:00
bb58435da0 Fix potential synchronization issues with framelisteners in EglRenderer.
Previously, a frame queued before calling addFrameListener could be
passed to the listener. Also fixes an issue where listener could still
be called after removeFrameListener call returned.

BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2529313002
Cr-Commit-Position: refs/heads/master@{#15275}
2016-11-28 16:53:50 +00:00
266f0a44eb Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
In this CL:
 - EndToEndTests is now parameterized.
 - Added VP8 non-rotated unittest.
 - CanReceiveUlpfec/CanReceiveFlexFec now use multisets for timestamps.
 - pre_decode_image_callback_ is now called before decoding a frame
   with the new video jitter buffer.
 - Set video rotation when FrameObjects are created.
 - Calculate KeyFramesReceivedInPermille in new video jitter buffer.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2522493002
Cr-Commit-Position: refs/heads/master@{#15274}
2016-11-28 16:49:15 +00:00
d1aaaaa125 Remove surface size mismatch logic from EglRenderer.
This logic doesn't really work. Application should mask the view while
the surface size is being changed.

BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2528243003
Cr-Commit-Position: refs/heads/master@{#15273}
2016-11-28 16:47:12 +00:00
6287e82b9b Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.

Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}

TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303

Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
2016-11-28 16:05:23 +00:00
7703b27c42 Disable PeerConnectionEndToEndTest.CallWithLegacySdp on Asan bots.
BUG=webrtc:6765
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2534783003 .

Cr-Commit-Position: refs/heads/master@{#15271}
2016-11-28 15:23:20 +00:00