Commit Graph

15875 Commits

Author SHA1 Message Date
91873b74c5 Roll chromium_revision 70957b2671..cf2dce6a6d (448581:448969)
Change log: 70957b2671..cf2dce6a6d
Full diff: 70957b2671..cf2dce6a6d

Changed dependencies:
* src/base: 3a1e5145d5..2a741233ff
* src/build: a5b79d9869..c4b2c5ac66
* src/ios: 533d4812cc..53837a325a
* src/testing: 908af2e604..2d16559171
* src/third_party: 8694363380..6549709ce7
* src/third_party/catapult: df67b47911..f20052d2ac
* src/third_party/ffmpeg: bc2eb1987e..239c9f9e27
* src/third_party/gtest-parallel: e281b59a8e..1dad0e9f6d
* src/tools: 9491f0f1c7..644b36a36a
DEPS diff: 70957b2671..cf2dce6a6d/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2684943002
Cr-Commit-Position: refs/heads/master@{#16493}
2017-02-08 13:49:35 +00:00
e525d6aba6 Revert Make the new jitter buffer the default jitter buffer.
Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2682073003
Cr-Commit-Position: refs/heads/master@{#16492}
2017-02-08 13:25:42 +00:00
498ee8e816 Remove repeat flag from SendRTCP
It is always false

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2684023002
Cr-Commit-Position: refs/heads/master@{#16491}
2017-02-08 13:24:31 +00:00
fd8f102a84 Revert of Avoid calling PostTask in audio callbacks (patchset #6 id:100001 of https://codereview.webrtc.org/2663383004/ )
Reason for revert:
Speculative revert to see if this CL caused a change in performance tests.

See https://bugs.chromium.org/p/chromium/issues/detail?id=689919 for details.

Original issue's description:
> Avoid calling PostTask in audio callbacks.
>
> We have seen that PostTask can consume some CPU and the way we used it
> before (logging only) in the ADB is not worth the cost we see when
> profiling.
>
> This CL simply moves frequent (trivial) stat updates from the task queue
> to the native threads to avoid calling PostTask in each callback.
> The reason for doing so before was to avoid locks but we can live without
> them since races are benign here.
>
>
> BUG=webrtc:7096
>
> Review-Url: https://codereview.webrtc.org/2663383004
> Cr-Commit-Position: refs/heads/master@{#16429}
> Committed: 77ce9a5541

TBR=solenberg@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7096

Review-Url: https://codereview.webrtc.org/2684913003
Cr-Commit-Position: refs/heads/master@{#16490}
2017-02-08 13:23:15 +00:00
219208991b Adding full initial version of delay estimation functionality in echo
canceller 3

This CL adds code to the all the delay estimation functionality that is
available for the first version of echo canceller 3. The code completes
the class EchoPathDelayEstimator.

Note that this code does not yet include any handling of clock-drift so
there will be upcoming versions of this code.

Also note that the CL includes some minor changes in other files for
echo canceller 3.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2644123002
Cr-Commit-Position: refs/heads/master@{#16489}
2017-02-08 13:08:56 +00:00
d4ed7f59e4 New tool for printing basic packet information from an RTC event log to stdout.
BUG=webrtc:7118

Review-Url: https://codereview.webrtc.org/2673403002
Cr-Commit-Position: refs/heads/master@{#16488}
2017-02-08 12:22:53 +00:00
abcef5d32e Replace std::tr1::tuple by ::testing::tuple.
BUG=webrtc:7129
NOTRY=True

Review-Url: https://codereview.webrtc.org/2686453004
Cr-Commit-Position: refs/heads/master@{#16487}
2017-02-08 12:07:11 +00:00
5e5a072f2c iOS: Fix breakage caused by buildbot recipe update
Similar changes as in https://codereview.chromium.org/2659123003
Our MB configuration is still the actually used build config for these
bots.

BUG=686304
TBR=smut@google.com

Review-Url: https://codereview.webrtc.org/2685813003 .
Cr-Commit-Position: refs/heads/master@{#16486}
2017-02-08 11:04:20 +00:00
b10f32f9b2 Adding more comments to every header file in api/ subdirectory.
Many of these interfaces are not intuitive, or are the way they are for
complex historical reasons, so it would be nice to document these things
for future developers.

Also, many nonstandard things (such as RTCConfiguration options) were
not documented at all before this CL.

BUG=webrtc:7131
TBR=pthatcher@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2680273002
Cr-Commit-Position: refs/heads/master@{#16485}
2017-02-08 09:38:21 +00:00
76e02cd4d8 Reland of ll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2680743003/ )
Reason for revert:
The downstream problem should be fixed now.

Original issue's description:
> Revert of Roll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2683593002/ )
>
> Reason for revert:
> Breaks internal projects.
>
> Original issue's description:
> > Roll chromium_revision 496a750d38..70957b2671 (447619:448581)
> >
> > Change log: 496a750d38..70957b2671
> > Full diff: 496a750d38..70957b2671
> >
> > Changed dependencies:
> > * src/base: 32f2a4543f..3a1e5145d5
> > * src/build: 337c73855e..a5b79d9869
> > * src/ios: 6b87d69c72..533d4812cc
> > * src/testing: 04c1f97a2d..908af2e604
> > * src/third_party: c9a58f7ae6..8694363380
> > * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/358baeb9a4..3f2611a98f
> > * src/third_party/catapult: a801abb6bc..df67b47911
> > * src/third_party/gtest-parallel: 8768563f5c..e281b59a8e
> > * src/third_party/libFuzzer/src: 78ee52d0c6..64bdf91585
> > * src/tools: d4ba547dba..9491f0f1c7
> > DEPS diff: 496a750d38..70957b2671/DEPS
> >
> > No update to Clang.
> >
> > TBR=
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2683593002
> > Cr-Commit-Position: refs/heads/master@{#16466}
> > Committed: 040f5cc5d7
>
> TBR=buildbot@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2680743003
> Cr-Commit-Position: refs/heads/master@{#16468}
> Committed: f81be0a83b

TBR=buildbot@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7126

Review-Url: https://codereview.webrtc.org/2687643003
Cr-Commit-Position: refs/heads/master@{#16484}
2017-02-08 09:27:33 +00:00
54b6e982df Added gn target for rtc_event_log2rtp_dump.
BUG=webrtc:6191

Review-Url: https://codereview.webrtc.org/2629353004
Cr-Commit-Position: refs/heads/master@{#16483}
2017-02-08 08:28:09 +00:00
7798501d7a Fix the Chrome crash caused by RtcEventLog
Stop the RtcEventLog when the PeerConnection is closed so that Chrome
will not crash because of creating too many threads.

BUG=chromium:687553

Review-Url: https://codereview.webrtc.org/2682433005
Cr-Commit-Position: refs/heads/master@{#16482}
2017-02-07 23:45:16 +00:00
9dd77baca4 Clarifying error messages in ParseIceServerUrl for invalid transport parameters.
BUG=webrtc:6662

Review-Url: https://codereview.webrtc.org/2680023005
Cr-Commit-Position: refs/heads/master@{#16481}
2017-02-07 23:09:50 +00:00
69fb2cca4d Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
Reason for revert:
Breaks downstream build.

Original issue's description:
> Add QP sum stats for received streams.
>
> This is not implemented yet in any of the decoders.
>
> BUG=webrtc:6541
>
> Review-Url: https://codereview.webrtc.org/2649133005
> Cr-Commit-Position: refs/heads/master@{#16475}
> Committed: ff0e72fd16

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,magjed@webrtc.org,stefan@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2680893002 .
Cr-Commit-Position: refs/heads/master@{#16480}
2017-02-07 18:59:25 +00:00
ed02c6d68f Revert of RTCInboundRTPStreamStats.qpSum collected. (patchset #4 id:80001 of https://codereview.webrtc.org/2675943002/ )
Reason for revert:
Breaks downstream build.

Original issue's description:
> RTCInboundRTPStreamStats.qpSum collected.
>
> This was previously only collected for local tracks
> (RTCOutboundRTPStreamStats.qpSum).
>
> Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
>
> This CL also improves some testing in rtcstatscollector_unittest.cc.
> Default and non-default values are tested in the same unittests,
> removing the test that was specific to default-values, which was
> otherwise code duplication.
>
> BUG=webrtc:7065
>
> Review-Url: https://codereview.webrtc.org/2675943002
> Cr-Commit-Position: refs/heads/master@{#16477}
> Committed: cd195bea5e

TBR=sakal@webrtc.org,hta@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7065

Review-Url: https://codereview.webrtc.org/2687483002 .
Cr-Commit-Position: refs/heads/master@{#16479}
2017-02-07 18:45:31 +00:00
76bc8e858f Delete VideoReceiveStream::Config::pre_render_callback.
Also delete the class I420FrameCallback.

BUG=webrtc:7124

Review-Url: https://codereview.webrtc.org/2678343002
Cr-Commit-Position: refs/heads/master@{#16478}
2017-02-07 17:37:41 +00:00
cd195bea5e RTCInboundRTPStreamStats.qpSum collected.
This was previously only collected for local tracks
(RTCOutboundRTPStreamStats.qpSum).

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum

This CL also improves some testing in rtcstatscollector_unittest.cc.
Default and non-default values are tested in the same unittests,
removing the test that was specific to default-values, which was
otherwise code duplication.

BUG=webrtc:7065

Review-Url: https://codereview.webrtc.org/2675943002
Cr-Commit-Position: refs/heads/master@{#16477}
2017-02-07 16:31:27 +00:00
c16fa5ea69 Replace all use of the VERIFY macro.
Replaced by assigning value to a local variable, followed by a DCHECK.
Also deletes dead test code under the always false TEST_DIGEST define.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2623473004
Cr-Commit-Position: refs/heads/master@{#16476}
2017-02-07 15:18:43 +00:00
ff0e72fd16 Add QP sum stats for received streams.
This is not implemented yet in any of the decoders.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2649133005
Cr-Commit-Position: refs/heads/master@{#16475}
2017-02-07 15:15:17 +00:00
7de8d64f89 Wire up audio packet loss to BWE.
BUG=webtrc:5079

Review-Url: https://codereview.webrtc.org/2658233002
Cr-Commit-Position: refs/heads/master@{#16474}
2017-02-07 15:14:08 +00:00
2bc6864278 Reland of Drop frames until specified bitrate is achieved. (patchset #1 id:1 of https://codereview.webrtc.org/2666303002/ )
Reason for revert:
Perf test broke as it made assumptions that quality scaling was turned off. This turns out not to be the case. Fixed by turning quality scaling off for the tests.

Original issue's description:
> Revert of Drop frames until specified bitrate is achieved. (patchset #12 id:240001 of https://codereview.webrtc.org/2630333002/ )
>
> Reason for revert:
> due to failures on perf tests (not on perf stats, but fails running due to dcheck failures), see e.g., https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20(K%20Nexus5)
>
> Original issue's description:
> > Drop frames until specified bitrate is achieved.
> >
> > This CL fixes a regression introduced with the new quality scaler
> > where the video would no longer start in a scaled mode. This CL adds
> > code that compares incoming captured frames to the target bitrate,
> > and if they are found to be too large, they are dropped and sinkWants
> > set to a lower resolution. The number of dropped frames should be low
> > (0-4 in most cases) and should not introduce a noticeable delay, or
> > at least should be preferrable to having the first 2-4 seconds of video
> > have very low quality.
> >
> > BUG=webrtc:6953
> >
> > Review-Url: https://codereview.webrtc.org/2630333002
> > Cr-Commit-Position: refs/heads/master@{#16391}
> > Committed: 83399caec5
>
> TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6953
>
> Review-Url: https://codereview.webrtc.org/2666303002
> Cr-Commit-Position: refs/heads/master@{#16395}
> Committed: 35fc2aa82f

TBR=perkj@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,minyue@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6953

Review-Url: https://codereview.webrtc.org/2675223002
Cr-Commit-Position: refs/heads/master@{#16473}
2017-02-07 15:02:22 +00:00
338f78ac95 RTCIceCandidatePairStats.available[Outgoing/Incoming]Bitrate collected.
Collected for current pairs, undefined for other pairs. This is the
same as the old stats' VideoBwe.googAvailable[Send/Receive]Bandwidth.

NOTE: The value this is based on for incoming bitrate is not set. This
CL wires it up but has a TODO that the incoming bitrate needs to be
collected properly. (Same problem for both old and new stats.)

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-availableoutgoingbitrate
Discussion: https://github.com/w3c/webrtc-stats/issues/112#issuecomment-277167781

BUG=webrtc:7062

Review-Url: https://codereview.webrtc.org/2675923002
Cr-Commit-Position: refs/heads/master@{#16472}
2017-02-07 14:41:21 +00:00
3443bb75a0 RTCRTPStreamStats.ssrc changed type to uint32_t.
As per PR: https://github.com/w3c/webrtc-stats/pull/157

BUG=webrtc:7065, webrtc:7066

Review-Url: https://codereview.webrtc.org/2675583003
Cr-Commit-Position: refs/heads/master@{#16471}
2017-02-07 14:28:11 +00:00
87b8e9f3a2 Add missing dependency to audio_decoder_unittests.
BUG=None
NOTRY=True
R=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2684573002
Cr-Commit-Position: refs/heads/master@{#16470}
2017-02-07 14:26:29 +00:00
a53d4e7b4f Reduce parallel jobs in build_aar.py to 200 when building with goma.
Previously we were using 1024 parallel jobs. This is too much when
running on bots.

BUG=webrtc:7023
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2679123002
Cr-Commit-Position: refs/heads/master@{#16469}
2017-02-07 14:19:20 +00:00
f81be0a83b Revert of Roll chromium_revision 496a750d38..70957b2671 (447619:448581) (patchset #1 id:1 of https://codereview.webrtc.org/2683593002/ )
Reason for revert:
Breaks internal projects.

Original issue's description:
> Roll chromium_revision 496a750d38..70957b2671 (447619:448581)
>
> Change log: 496a750d38..70957b2671
> Full diff: 496a750d38..70957b2671
>
> Changed dependencies:
> * src/base: 32f2a4543f..3a1e5145d5
> * src/build: 337c73855e..a5b79d9869
> * src/ios: 6b87d69c72..533d4812cc
> * src/testing: 04c1f97a2d..908af2e604
> * src/third_party: c9a58f7ae6..8694363380
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/358baeb9a4..3f2611a98f
> * src/third_party/catapult: a801abb6bc..df67b47911
> * src/third_party/gtest-parallel: 8768563f5c..e281b59a8e
> * src/third_party/libFuzzer/src: 78ee52d0c6..64bdf91585
> * src/tools: d4ba547dba..9491f0f1c7
> DEPS diff: 496a750d38..70957b2671/DEPS
>
> No update to Clang.
>
> TBR=
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2683593002
> Cr-Commit-Position: refs/heads/master@{#16466}
> Committed: 040f5cc5d7

TBR=buildbot@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2680743003
Cr-Commit-Position: refs/heads/master@{#16468}
2017-02-07 13:48:12 +00:00
585a9b191c Refactor and clean-up relating to RTCCodecStats.
Refactor how |codec_id| is set, remove outdated TODO, update comments
with new bugs IDs.

BUG=webrtc:7061

Review-Url: https://codereview.webrtc.org/2670343002
Cr-Commit-Position: refs/heads/master@{#16467}
2017-02-07 12:59:16 +00:00
040f5cc5d7 Roll chromium_revision 496a750d38..70957b2671 (447619:448581)
Change log: 496a750d38..70957b2671
Full diff: 496a750d38..70957b2671

Changed dependencies:
* src/base: 32f2a4543f..3a1e5145d5
* src/build: 337c73855e..a5b79d9869
* src/ios: 6b87d69c72..533d4812cc
* src/testing: 04c1f97a2d..908af2e604
* src/third_party: c9a58f7ae6..8694363380
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/358baeb9a4..3f2611a98f
* src/third_party/catapult: a801abb6bc..df67b47911
* src/third_party/gtest-parallel: 8768563f5c..e281b59a8e
* src/third_party/libFuzzer/src: 78ee52d0c6..64bdf91585
* src/tools: d4ba547dba..9491f0f1c7
DEPS diff: 496a750d38..70957b2671/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2683593002
Cr-Commit-Position: refs/heads/master@{#16466}
2017-02-07 12:54:19 +00:00
b99b5960e1 Add chromium-junit4 tag to instrumentation test AndroidManifests.
This tag is supposed to be temporary and removed when all Chromium tests
have been migrated to JUnit4.

BUG=webrtc:7123,chromium:640116
NOTRY=True

Review-Url: https://codereview.webrtc.org/2683583002
Cr-Commit-Position: refs/heads/master@{#16465}
2017-02-07 12:12:30 +00:00
e0ac5a6c15 Use std::unique_ptr in VideoProcessorIntegrationTest.
Add more logging of codec settings.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2639203005
Cr-Commit-Position: refs/heads/master@{#16464}
2017-02-07 11:54:04 +00:00
1b21b9bbf8 Replace occurences of string by std::string.
BUG=None
NOTRY=True

Review-Url: https://codereview.webrtc.org/2685533002
Cr-Commit-Position: refs/heads/master@{#16463}
2017-02-07 11:40:28 +00:00
1634e16042 Remove use of selectors matching Apple private API names.
This was causing some apps that include WebRTC to be rejected from the
app store.

BUG=webrtc:6382

Review-Url: https://codereview.webrtc.org/2679913002
Cr-Commit-Position: refs/heads/master@{#16462}
2017-02-07 10:48:55 +00:00
4a9a595ab4 Make rtcp packets copyable
That would simplify their usage in tests where perfomance is not critical.

BUG=None

Review-Url: https://codereview.webrtc.org/2675713005
Cr-Commit-Position: refs/heads/master@{#16461}
2017-02-07 09:53:04 +00:00
1959b63b61 Remove Assert lint suppression.
BUG=webrtc:6597
NOTRY=True

Review-Url: https://codereview.webrtc.org/2668963002
Cr-Commit-Position: refs/heads/master@{#16460}
2017-02-07 09:41:47 +00:00
4709e8971b Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
We can then drop the CongestionController and RemoteBitrateEstimator
completely from the receive streams.

BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2669463006
Cr-Commit-Position: refs/heads/master@{#16459}
2017-02-07 09:18:43 +00:00
6b3fcfd823 Add support for extra GN args to Android build script.
By using the --extra-gn-args flag, it is now possible to
specify additional GN arguments for the build. This is needed
in order to pass a non-default Goma directory (needed for the bots).
Example use: --extra-gn-args goma_dir=\"/path/to/goma\"
You can also pass multiple args (separated by spaces).

BUG=chromium:684387
NOTRY=True
TESTED=Did a local successful run.

Review-Url: https://codereview.webrtc.org/2670743004
Cr-Commit-Position: refs/heads/master@{#16458}
2017-02-07 09:11:06 +00:00
6b34124a6d Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/
BUG=webrtc:6847

Review-Url: https://codereview.webrtc.org/2663063008
Cr-Commit-Position: refs/heads/master@{#16457}
2017-02-06 21:39:38 +00:00
f748ca47b0 Change order of tear down/create of default audio stream, to avoid starting/stopping audio card playout unnecessarily.
BUG=b/34746131

Review-Url: https://codereview.webrtc.org/2670183005
Cr-Commit-Position: refs/heads/master@{#16456}
2017-02-06 21:03:19 +00:00
bd9a77f4e5 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.
(TBRing webrtc/test/ OWNER)

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

Review-Url: https://codereview.webrtc.org/2669153004
Cr-Commit-Position: refs/heads/master@{#16455}
2017-02-06 20:53:57 +00:00
f9b6e5e4b5 Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations.
BUG=webrtc:7111
R=brandtr@webrtc.org

Review-Url: https://codereview.webrtc.org/2676973004 .
Cr-Commit-Position: refs/heads/master@{#16454}
2017-02-06 16:17:57 +00:00
7a2d8ca9bc Rewrite iOS FAT libraries build script in Python
BUG=webrtc:7049

Review-Url: https://codereview.webrtc.org/2662513004
Cr-Commit-Position: refs/heads/master@{#16453}
2017-02-06 15:53:41 +00:00
1134b7b918 Reland of Improve and re-enable FEC end-to-end tests. (patchset #1 id:1 of https://codereview.webrtc.org/2672373002/ )
Reason for revert:
Will try to reland FlexFEC tests, since these do not seem to be flaky on the buildbots.

Original issue's description:
> Revert of Improve and re-enable FEC end-to-end tests. (patchset #3 id:40001 of https://codereview.webrtc.org/2675573004/ )
>
> Reason for revert:
> Ulpfec tests are still flaky on buildbots.
>
> Original issue's description:
> > Improve and re-enable FEC end-to-end tests.
> >
> > These tests got flaky under the new jitter buffer.
> >
> > Enhancements:
> > - Use send-side BWE.
> > - Let BWE ramp up before applying packet loss.
> > - Improve packet loss simulation for ULPFEC.
> > - Add delay to fake network pipe for FlexFEC.
> >   (Not added for ULPFEC, since this makes those flaky...?)
> > - Add FlexFEC+NACK test, using RTX instead of "raw retransmits".
> > - Tighter checks of received packets' payload types and SSRCs.
> >
> > TESTED=
> > $ ninja -C out/Debug video_engine_tests && third_party/gtest-parallel/gtest-parallel -r 1000 out/Debug/video_engine_tests --gtest_filter="*EndToEnd*Ulpfec*:*EndToEnd*Flexfec*"
> > ninja: Entering directory `out/Debug'
> > ninja: no work to do.
> > [12000/12000] TestWithNewVideoJitterBuffer/EndToEndTest.RecoversWithFlexfecAndNack/1 (14935 ms)
> >
> > BUG=webrtc:7047
> >
> > Review-Url: https://codereview.webrtc.org/2675573004
> > Cr-Commit-Position: refs/heads/master@{#16449}
> > Committed: d40b0f39e0
>
> TBR=stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7047
>
> Review-Url: https://codereview.webrtc.org/2672373002
> Cr-Commit-Position: refs/heads/master@{#16450}
> Committed: fd8d2654d7

TBR=stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7047

Review-Url: https://codereview.webrtc.org/2675283003
Cr-Commit-Position: refs/heads/master@{#16452}
2017-02-06 14:35:47 +00:00
b77c716d8a Enable send-side BWE by default for video in call tests.
Also fixes a bug where RTCP transport feedback was sent even though RTCP was disabled.

May affect perf numbers since the behavior of the send-side BWE differs a lot from the recv-side BWE.

BUG=webrtc:7111

Review-Url: https://codereview.webrtc.org/2669413003
Cr-Commit-Position: refs/heads/master@{#16451}
2017-02-06 14:29:38 +00:00
fd8d2654d7 Revert of Improve and re-enable FEC end-to-end tests. (patchset #3 id:40001 of https://codereview.webrtc.org/2675573004/ )
Reason for revert:
Ulpfec tests are still flaky on buildbots.

Original issue's description:
> Improve and re-enable FEC end-to-end tests.
>
> These tests got flaky under the new jitter buffer.
>
> Enhancements:
> - Use send-side BWE.
> - Let BWE ramp up before applying packet loss.
> - Improve packet loss simulation for ULPFEC.
> - Add delay to fake network pipe for FlexFEC.
>   (Not added for ULPFEC, since this makes those flaky...?)
> - Add FlexFEC+NACK test, using RTX instead of "raw retransmits".
> - Tighter checks of received packets' payload types and SSRCs.
>
> TESTED=
> $ ninja -C out/Debug video_engine_tests && third_party/gtest-parallel/gtest-parallel -r 1000 out/Debug/video_engine_tests --gtest_filter="*EndToEnd*Ulpfec*:*EndToEnd*Flexfec*"
> ninja: Entering directory `out/Debug'
> ninja: no work to do.
> [12000/12000] TestWithNewVideoJitterBuffer/EndToEndTest.RecoversWithFlexfecAndNack/1 (14935 ms)
>
> BUG=webrtc:7047
>
> Review-Url: https://codereview.webrtc.org/2675573004
> Cr-Commit-Position: refs/heads/master@{#16449}
> Committed: d40b0f39e0

TBR=stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7047

Review-Url: https://codereview.webrtc.org/2672373002
Cr-Commit-Position: refs/heads/master@{#16450}
2017-02-06 14:19:51 +00:00
d40b0f39e0 Improve and re-enable FEC end-to-end tests.
These tests got flaky under the new jitter buffer.

Enhancements:
- Use send-side BWE.
- Let BWE ramp up before applying packet loss.
- Improve packet loss simulation for ULPFEC.
- Add delay to fake network pipe for FlexFEC.
  (Not added for ULPFEC, since this makes those flaky...?)
- Add FlexFEC+NACK test, using RTX instead of "raw retransmits".
- Tighter checks of received packets' payload types and SSRCs.

TESTED=
$ ninja -C out/Debug video_engine_tests && third_party/gtest-parallel/gtest-parallel -r 1000 out/Debug/video_engine_tests --gtest_filter="*EndToEnd*Ulpfec*:*EndToEnd*Flexfec*"
ninja: Entering directory `out/Debug'
ninja: no work to do.
[12000/12000] TestWithNewVideoJitterBuffer/EndToEndTest.RecoversWithFlexfecAndNack/1 (14935 ms)

BUG=webrtc:7047

Review-Url: https://codereview.webrtc.org/2675573004
Cr-Commit-Position: refs/heads/master@{#16449}
2017-02-06 13:54:43 +00:00
cb789bb510 Remove NewApi lint suppression.
BUG=webrtc:6597

Review-Url: https://codereview.webrtc.org/2662273004
Cr-Commit-Position: refs/heads/master@{#16448}
2017-02-06 13:34:26 +00:00
93e1e23537 Use RateAccCounter for sent bitrate stats. Reports average of periodically computed stats over a call.
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.BitrateSentInKbps"
"WebRTC.Video.MediaBitrateSentInKbps"
"WebRTC.Video.PaddingBitrateSentInKbps"
"WebRTC.Video.RetransmittedBitrateSentInKbps"
"WebRTC.Video.RtxBitrateSentInKbps"
"WebRTC.Video.FecBitrateSentInKbps"

BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2536613002
Cr-Commit-Position: refs/heads/master@{#16447}
2017-02-06 13:18:35 +00:00
447dba9586 Add debuggable=true to AppRTCMobile manifest.
Improves possibility to debug the demo application using adb.
As an example, now allows 'adb run as' which is useful when profiling CPU. It enables us to do profiling on non-rooted devices (excluding details).

BUG=NONE

Review-Url: https://codereview.webrtc.org/2676983003
Cr-Commit-Position: refs/heads/master@{#16446}
2017-02-06 12:58:01 +00:00
b114e9c159 Camera2Session: Add return statements after reportError where needed.
BUG=webrtc:7117

Review-Url: https://codereview.webrtc.org/2674243002
Cr-Commit-Position: refs/heads/master@{#16445}
2017-02-06 12:55:21 +00:00
873fcb958f Drop the check for stray mobileprovision (no longer needed)
BUG=webrtc:7049

Review-Url: https://codereview.webrtc.org/2676233002
Cr-Commit-Position: refs/heads/master@{#16444}
2017-02-06 11:43:58 +00:00