14613 Commits

Author SHA1 Message Date
magjed
5c71166dff VP8DecoderImpl: Fix uninitialized memory crash
It is not safe to call vpx_codec_destroy if vpx_codec_dec_init failed,
because the |decoder_| memory will be uninitialized. See the bug for
more info.

BUG=chromium:663293

Review-Url: https://codereview.webrtc.org/2541163007
Cr-Commit-Position: refs/heads/master@{#15381}
2016-12-02 10:46:26 +00:00
ossu
00bceb1eda Deprecated SetAudioPacketSize from RTPSender and removed calls to it.
The packet size was only used to control how often to output DTMF
packets. However, it likely did not work as intended, since that
interval was only set during initialization. No changes to the packet
size, like what AudioEncoder::Num10MsFramesInNextPacket could
indicate, were picked up. The value was instead taken from an entry in
ACMCodecDB.

Since it was not-fully-functional, its exact value didn't seem to
matter and it was getting in the way of making it possible to supply
an external audio encoder factory, I've decided to remove it
altogether. The DTMF code now uses an interval of 50 ms regardless,
which is a value recommended by the RFC.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2545753002
Cr-Commit-Position: refs/heads/master@{#15380}
2016-12-02 10:40:12 +00:00
henrik.lundin
e066b302ab Remove API-related #defines from voice_engine_configurations.h
BUG=webrtc:6506
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device;master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng

Review-Url: https://codereview.webrtc.org/2549443002
Cr-Commit-Position: refs/heads/master@{#15379}
2016-12-02 10:30:23 +00:00
buildbot
15cf9f92ac Roll chromium_revision 601e4f48a3..f50152dfc4 (435870:435897)
Change log: 601e4f48a3..f50152dfc4
Full diff: 601e4f48a3..f50152dfc4

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2548793002
Cr-Commit-Position: refs/heads/master@{#15378}
2016-12-02 10:07:50 +00:00
kthelgason
b336392562 Sanity check parsed QP values from H264 bitstream
BUG=chromium:663610

Review-Url: https://codereview.webrtc.org/2532973002
Cr-Commit-Position: refs/heads/master@{#15377}
2016-12-02 09:29:53 +00:00
buildbot
864f58b751 Roll chromium_revision 353f713f3d..601e4f48a3 (435846:435870)
Change log: 353f713f3d..601e4f48a3
Full diff: 353f713f3d..601e4f48a3

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2545753004
Cr-Commit-Position: refs/heads/master@{#15376}
2016-12-02 07:06:24 +00:00
buildbot
ea3d923bc8 Roll chromium_revision c6d437f401..353f713f3d (435797:435846)
Change log: c6d437f401..353f713f3d
Full diff: c6d437f401..353f713f3d

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2548463004
Cr-Commit-Position: refs/heads/master@{#15375}
2016-12-02 04:09:33 +00:00
buildbot
51cb31c468 Roll chromium_revision 47f73d2d11..c6d437f401 (435727:435797)
Change log: 47f73d2d11..c6d437f401
Full diff: 47f73d2d11..c6d437f401

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2549503005
Cr-Commit-Position: refs/heads/master@{#15374}
2016-12-02 01:07:07 +00:00
deadbeef
b465980fd7 In end-to-end PeerConnection tests, allow video to be downscaled.
QualityScaler may scale down the resolution, so our tests shouldn't
expect the input resolution to exactly match the resolution received on
the other side. Instead, we now just check that the aspect ratio
matches.

BUG=webrtc:5907

Review-Url: https://codereview.webrtc.org/2547673002
Cr-Commit-Position: refs/heads/master@{#15373}
2016-12-02 00:23:36 +00:00
buildbot
897530eb48 Roll chromium_revision bffc40ae72..47f73d2d11 (435660:435727)
Change log: bffc40ae72..47f73d2d11
Full diff: bffc40ae72..47f73d2d11

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2544923002
Cr-Commit-Position: refs/heads/master@{#15372}
2016-12-01 22:07:07 +00:00
deadbeef
8f89bff9a6 Revert of Disabled flaky P2PTestConductor tests on ASAN and MSAN. (patchset #1 id:1 of https://codereview.webrtc.org/2539103002/ )
Reason for revert:
The flaky tests should now be fixed.

Original issue's description:
> Disabled flaky P2PTestConductor tests on ASAN and MSAN.
>
> TBR=deadbeef@webrtc.org
> BUG=webrtc:6776
>
> Committed: https://crrev.com/8d66a5a3b18eef73b238f4220477da265bf4494b
> Cr-Commit-Position: refs/heads/master@{#15324}

TBR=ossu@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6776

Review-Url: https://codereview.webrtc.org/2550453003
Cr-Commit-Position: refs/heads/master@{#15371}
2016-12-01 20:54:28 +00:00
deadbeef
c6b6e09d18 Relaxing timeouts for TestMediaMonitor.
This isn't a performance test, so it may be running in a slow
environment, and shouldn't be subject to strict timeouts.

BUG=webrtc:6801
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2539183005
Cr-Commit-Position: refs/heads/master@{#15370}
2016-12-01 20:49:25 +00:00
deadbeef
8f425f9629 Relaxing DCHECK for packets sent before SRTP is enabled.
We still DCHECK for RTP, but not RTCP. RTCP packets can be sent before
offer/answer negotiation is complete, due to this bug:
https://bugs.chromium.org/p/webrtc/issues/detail?id=6809

This bug can only occur if the RTCP mux policy is "require", which is
why we started hitting it recently (the default in unit tests was
recently changed to "require").

BUG=webrtc:6776
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2542233002
Cr-Commit-Position: refs/heads/master@{#15369}
2016-12-01 20:26:33 +00:00
buildbot
183d51a67f Roll chromium_revision 3045382d44..bffc40ae72 (435618:435660)
Change log: 3045382d44..bffc40ae72
Full diff: 3045382d44..bffc40ae72

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2542813003
Cr-Commit-Position: refs/heads/master@{#15368}
2016-12-01 19:36:44 +00:00
henrik.lundin
f29e05d774 Add linearly spaced counting histograms
This change adds HistogramFactoryGetCountsLinear and
RTC_HISTOGRAM_COUNTS_LINEAR. Note that the default implementation of
HistogramFactoryGetCounts in metrics_default.cc also provides a
linearly spaced histogram, while the Chrome UMA implementation
provides exponentially spaced buckets.

BUG=none

Review-Url: https://codereview.webrtc.org/2548463002
Cr-Commit-Position: refs/heads/master@{#15367}
2016-12-01 17:58:53 +00:00
danilchap
1454669c1d Cleanup RtpHeaderExtensionMap removing use of two legacy functions
BUG=webrtc:1994

Review-Url: https://codereview.webrtc.org/2491273002
Cr-Commit-Position: refs/heads/master@{#15366}
2016-12-01 16:39:44 +00:00
buildbot
73f2ee2556 Roll chromium_revision a6a331af3e..3045382d44 (435597:435618)
Change log: a6a331af3e..3045382d44
Full diff: a6a331af3e..3045382d44

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2541383002
Cr-Commit-Position: refs/heads/master@{#15365}
2016-12-01 16:17:10 +00:00
terelius
182e4a4aff Remove bitrate cap for AdaptiveVideoSource and increase other caps to 25 Mbps.
AdaptiveVideoSource is used in testing/simulations of the bandwidth estimator.

Nada's reaction to delay depends on the current bitrate and the configured max rate in a non-intuituve way. Increase the starting bitrate to compensate for the increased max bitrate. This is only used in unit tests.

BUG=webrtc:6807

# Presubmit warns about a lint error in bwe.h that's unrelated to my change. Fixing it is beyond the scope of this CL.
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2542843003
Cr-Commit-Position: refs/heads/master@{#15364}
2016-12-01 15:29:15 +00:00
tandrii
2d9c877c64 Revert of Whitespace.
(patchset #1 id:1 of https://codereview.webrtc.org/2547543002/ )

Reason for revert:
whitespace reverts need no reason.

Original issue's description:
> Whitespace.
>
> BUG=668624
> TBR=kjellander@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/3b56e1f7b344e3a48b6765962f2e621727ef1519
> Cr-Commit-Position: refs/heads/master@{#15360}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=668624

Review-Url: https://codereview.webrtc.org/2549523002
Cr-Commit-Position: refs/heads/master@{#15363}
2016-12-01 15:13:07 +00:00
Andrii Shyshkalov
106edaeb85 Revert of Whitespace.
(patchset #1 id:1 of https://codereview.webrtc.org/2547553002/)

Reason for revert:
whitespace reverts need no reason.

Original issue's description:
> Whitespace.
>
> For manual commit.
>
> BUG=668624
> TBR=kjellander@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/aa7a9a2dc03c11ea6dc52a8424c4485e2f48d650
> Cr-Commit-Position: refs/heads/master@{#15361}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=668624

Review URL: https://codereview.webrtc.org/2549453003 .

Cr-Commit-Position: refs/heads/master@{#15362}
2016-12-01 15:10:21 +00:00
Andrii Shyshkalov
aa7a9a2dc0 Whitespace.
For manual commit.

BUG=668624
TBR=kjellander@webrtc.org
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/2547553002 .

Cr-Commit-Position: refs/heads/master@{#15361}
2016-12-01 15:07:13 +00:00
tandrii
3b56e1f7b3 Whitespace.
BUG=668624
TBR=kjellander@webrtc.org
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2547543002
Cr-Commit-Position: refs/heads/master@{#15360}
2016-12-01 15:05:54 +00:00
sprang
1a646ee522 Wire up BitrateAllocation to be sent as RTCP TargetBitrate
This is the video parts of https://codereview.webrtc.org/2531383002/
Wire up BitrateAllocation to be sent as RTCP TargetBitrate

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2541303003
Cr-Commit-Position: refs/heads/master@{#15359}
2016-12-01 14:34:18 +00:00
sprang
5e38c967e0 Wire up RTCP XR target bitrate in rtp/rtcp module
This is breakout of the rtcp parts of
https://codereview.webrtc.org/2531383002/

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2546713002
Cr-Commit-Position: refs/heads/master@{#15358}
2016-12-01 13:18:19 +00:00
buildbot
b57e0f6836 Roll chromium_revision 698c4714e8..a6a331af3e (435589:435597)
Change log: 698c4714e8..a6a331af3e
Full diff: 698c4714e8..a6a331af3e

No dependencies changed.
No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2537303005
Cr-Commit-Position: refs/heads/master@{#15357}
2016-12-01 13:08:54 +00:00
kthelgason
5e13d41124 Remove limit on how often quality scaling downscales
When starting from 720p this is necessary to achieve acceptable
quality at low bitrates.

BUG=webrtc:6495

Review-Url: https://codereview.webrtc.org/2538913003
Cr-Commit-Position: refs/heads/master@{#15356}
2016-12-01 11:59:56 +00:00
kthelgason
86cf9a2474 Increase test timeout to combat flakiness.
These tests have been a little flaky on the bots.
Hopefully increasing the timeout by 200% will help.

BUG=webrtc:6799

Review-Url: https://codereview.webrtc.org/2541743006
Cr-Commit-Position: refs/heads/master@{#15355}
2016-12-01 10:57:07 +00:00
mflodman
e90adcef42 Remove OnLocalSsrcChanged
Removing the unused interface OnLocalSsrcChanged.

BUG=webrtc:6456

Review-Url: https://codereview.webrtc.org/2546703002
Cr-Commit-Position: refs/heads/master@{#15354}
2016-12-01 10:39:49 +00:00
buildbot
847f2948e5 Roll chromium_revision b5ce7e0a17..698c4714e8 (435129:435589)
Change log: b5ce7e0a17..698c4714e8
Full diff: b5ce7e0a17..698c4714e8

Changed dependencies:
* src/buildtools: 991f459071..102c16366d
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/e1cc35e581..f086df9f5f
DEPS diff: b5ce7e0a17..698c4714e8/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2539413002
Cr-Commit-Position: refs/heads/master@{#15353}
2016-12-01 10:23:53 +00:00
magjed
665bc3c7ad Move webrtc/api/androidtests to webrtc/sdk/android/instrumentationtests
BUG=webrtc:5882
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2541823002
Cr-Commit-Position: refs/heads/master@{#15352}
2016-12-01 09:45:35 +00:00
kthelgason
2bc324cc5a Add method on AVFoundation capturer to adapt output format.
This CL makes a method available on the AVFoundationVideoCapturer
that adapts the output format of captured video to the specified
width and height.

BUG=webrtc:6753

Review-Url: https://codereview.webrtc.org/2528493004
Cr-Commit-Position: refs/heads/master@{#15351}
2016-12-01 09:36:22 +00:00
magjed
dd40702357 Move VideoDecoder::Create() logic to separate internal video decoder factory
The goal with this CL is to move implementation details out from the
webrtc root (webrtc/video_decoder.h) to simplify the dependency graph.
Another goal is to streamline the creation of VideoDecoders in
webrtcvideoengine2.cc; it will now have two factories of the same
WebRtcVideoDecoderFactory type, one internal and one external.

Specifically, this CL:
 * Removes webrtc::VideoDecoder::DecoderType and use webrtc::VideoCodecType
   instead.
 * Removes 'static VideoDecoder* Create(DecoderType codec_type)' and
   moves the create function to the internal decoder factory instead.
 * Removes video_decoder.cc. webrtc::VideoDecoder is now just an
   interface without any static functions.

BUG=webrtc:6743

Review-Url: https://codereview.webrtc.org/2521203002
Cr-Commit-Position: refs/heads/master@{#15350}
2016-12-01 08:27:35 +00:00
brandtr
aa354c9512 Rename full_stack.cc to full_stack_tests.cc.
Also rename the accompanying plot file.

BUG=None

Review-Url: https://codereview.webrtc.org/2529293006
Cr-Commit-Position: refs/heads/master@{#15349}
2016-12-01 08:20:24 +00:00
kthelgason
a974d76c74 Enable VideoToolbox encoder on mac
BUG=webrtc:6317

Review-Url: https://codereview.webrtc.org/2532983006
Cr-Commit-Position: refs/heads/master@{#15348}
2016-12-01 08:16:54 +00:00
Henrik Kjellander
024c90ed90 Whitespace CL to trigger bots.
TBR=ehmaldonado@webrtc.org
BUG=None

Review URL: https://codereview.webrtc.org/2547453002 .

Cr-Commit-Position: refs/heads/master@{#15347}
2016-12-01 07:05:44 +00:00
Henrik Kjellander
3a70cc3977 Whitespace change to trigger bots.
TBR=ehmaldonado@webrtc.org
BUG=None

Review URL: https://codereview.webrtc.org/2541323002 .

Cr-Commit-Position: refs/heads/master@{#15346}
2016-12-01 06:04:11 +00:00
buildbot
5c13c33f30 Roll chromium_revision b66d8ae9dc..b5ce7e0a17 (435081:435129)
Change log: b66d8ae9dc..b5ce7e0a17
Full diff: b66d8ae9dc..b5ce7e0a17

Changed dependencies:
* src/third_party/ffmpeg: d16162e3f4..7e5307d753
DEPS diff: b66d8ae9dc..b5ce7e0a17/DEPS

No update to Clang.

TBR=
BUG=None

Review-Url: https://codereview.webrtc.org/2529293005
Cr-Commit-Position: refs/heads/master@{#15345}
2016-12-01 00:14:47 +00:00
skvlad
ed6e077263 Make SurfaceTextureHelper and I420Frame public in Java.
This change makes the Java classes and constructors for
SurfaceTextureHelper and I420Frame public. This allows applications to
use the WebRTC CameraVideoCapturer to obtain raw frames, and to render
frames on the WebRTC VideoRenderer without having to pass them through a
VideoTrack - such as when using Quartc.

BUG=None.

Review-Url: https://codereview.webrtc.org/2544563002
Cr-Commit-Position: refs/heads/master@{#15344}
2016-11-30 22:40:23 +00:00
terelius
a15948c9c6 Change assert to RTC_DCHECK in bwe_test_logging.cc
BUG=None
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2546483002
Cr-Commit-Position: refs/heads/master@{#15343}
2016-11-30 17:02:26 +00:00
sakal
3a9bc1790a Allow custom drawers to be added to framelisteners.
BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2541053002
Cr-Commit-Position: refs/heads/master@{#15342}
2016-11-30 16:30:11 +00:00
henrik.lundin
06c1d6484e Prep to remove API-related #defines from voice_engine_configurations.h
The follwing #defines are marked as deprecated:
- WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- WEBRTC_VOICE_ENGINE_HARDWARE_API
- WEBRTC_VOICE_ENGINE_FILE_API
- WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- WEBRTC_VOICE_ENGINE_CODEC_API
- WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API

BUG=webrtc:6506

Review-Url: https://codereview.webrtc.org/2538093003
Cr-Commit-Position: refs/heads/master@{#15341}
2016-11-30 16:21:54 +00:00
michaelt
9332b7d0ad Reland "Update rtt on audio only calls".
https://codereview.webrtc.org/2402333002

BUG=webrtc:6508

Review-Url: https://codereview.webrtc.org/2530383002
Cr-Commit-Position: refs/heads/master@{#15340}
2016-11-30 15:51:19 +00:00
brandtr
93c5d030fc Start gathering perf data for VP8 + FlexFEC.
This is to assess the performance penalty of the (current)
lack of integration with FlexFEC and BWE.

This CL also enables send-side BWE for the following tests:
- foreman_cif_net_delay_0_0_plr_0_VP9
- foreman_cif_net_delay_0_0_plr_0_H264
- foreman_cif_delay_50_0_plr_5_VP9
- foreman_cif_delay_50_0_plr_5_H264
- foreman_cif_delay_50_0_plr_5_H264_flexfec
- foreman_cif_delay_50_0_plr_5_H264_ulpfec
Perf alerts on these tests are therefore expected.

R=stefan@webrtc.org
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2534203004
Cr-Commit-Position: refs/heads/master@{#15339}
2016-11-30 15:50:13 +00:00
nisse
13d38fbe90 Delete all of the video_processing module but the denoiser code.
It is unused since cl https://codereview.webrtc.org/2386573002.

The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.

BUG=None

Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
2016-11-30 15:44:59 +00:00
stefan
13f1a0a9ca Wire up x-google-{min,start,max}-bitrate to WebRtcVoiceMediaChannel.
BUG=webrtc:6793

Review-Url: https://codereview.webrtc.org/2534173002
Cr-Commit-Position: refs/heads/master@{#15337}
2016-11-30 15:23:07 +00:00
kthelgason
127387e5ce Delete nalu parser in mediaencoder
We already have an implementation in h264_common. We should have
as few of these as possible as they are subtly hard to get right
and it creates work to maintain N implementations.

BUG=webrtc:6546

Review-Url: https://codereview.webrtc.org/2538133002
Cr-Commit-Position: refs/heads/master@{#15336}
2016-11-30 15:12:54 +00:00
kthelgason
2305b701ae Remove unused build override
BUG=webrtc:6431

Review-Url: https://codereview.webrtc.org/2545503002
Cr-Commit-Position: refs/heads/master@{#15335}
2016-11-30 15:01:24 +00:00
terelius
cb861e074a Templatize percentile_filter.h and move it to base/analytics.
BUG=None

Review-Url: https://codereview.webrtc.org/2529063002
Cr-Commit-Position: refs/heads/master@{#15334}
2016-11-30 14:52:06 +00:00
minyue
4b9a2cb0d8 Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
The earlier attempt of this was in
https://codereview.webrtc.org/2411613002/

It was reverted due to failures on internal bots, showing that we cannot deprecate one method.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2538493006
Cr-Commit-Position: refs/heads/master@{#15333}
2016-11-30 14:50:08 +00:00
ehmaldonado
26bddb92f0 Replace test_support_main by test_main and get rid of test_support_main_threaded_mac
test_support_main_threaded_mac doesn't seem to be used. It looks like it was
last used about a year and a half ago, and was removed in
https://webrtc-codereview.appspot.com/55379004

BUG=webrtc:6424
R=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2540693002
Cr-Commit-Position: refs/heads/master@{#15332}
2016-11-30 14:12:10 +00:00