Commit Graph

11283 Commits

Author SHA1 Message Date
7f067663ac Delete deprecated PeerConnection methods, and corresponding using declarations.
BUG=None

Review-Url: https://codereview.webrtc.org/2632203003
Cr-Commit-Position: refs/heads/master@{#17120}
2017-03-08 14:59:45 +00:00
cbbd8c76e8 Revert of Add Metal video view in AppRTCMobile and metal availability macro. (patchset #5 id:80001 of https://codereview.webrtc.org/2722583002/ )
Reason for revert:
Breaks AppRTCMobile

Original issue's description:
> Add Metal video view in AppRTCMobile and Metal availability macro.
>
> - The RTC_SUPPORTS_METAL macro allows consumers to gracefully handle compilation for different archs that are not supporting Metal.
>
> BUG=webrtc:7079
>
> Review-Url: https://codereview.webrtc.org/2722583002
> Cr-Commit-Position: refs/heads/master@{#17004}
> Committed: 154a7bb877

TBR=magjed@webrtc.org,tkchin@webrtc.org,denicija@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7079

Review-Url: https://codereview.webrtc.org/2739793003
Cr-Commit-Position: refs/heads/master@{#17119}
2017-03-08 14:54:59 +00:00
96d91524fa Revert of Add unit tests for RTCMTLVideoView. (patchset #6 id:100001 of https://codereview.webrtc.org/2723903003/ )
Reason for revert:
This CL depends on a reverted CL.

Original issue's description:
> Add unit tests for RTCMTLVideoView.
>
> To properly test the functionality,  following changes  were needed
> - Make RTCMTLVideoView compiliable for all cpu architectures not just arm64.
> This is needed so that the test can run on any device and on simulator as well.
> - Refactor RTCMTLVideoView to have  mockable class methods.
> The unittest class, RTCMTLVideoViewTests was designed to provide easy transition
> to XCTest when the time comes for that.
> To transition to XCTest it would suffice to inherit from XCTestCase and remove
> the gtest methods.
>
> BUG=webrtc:7079
>
> Review-Url: https://codereview.webrtc.org/2723903003
> Cr-Commit-Position: refs/heads/master@{#17014}
> Committed: 0ebe0199ac

TBR=magjed@webrtc.org,denicija@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7079

Review-Url: https://codereview.webrtc.org/2733953006
Cr-Commit-Position: refs/heads/master@{#17118}
2017-03-08 14:33:52 +00:00
1d395dfb36 Conversational Speech generator, main script with shell arguments
BUG=webrtc:7218
NOTRY=True

Review-Url: https://codereview.webrtc.org/2733863002
Cr-Commit-Position: refs/heads/master@{#17117}
2017-03-08 14:12:23 +00:00
a8d8aadba8 Refactor + enable GN check on video_coding_utility
To avoid the cyclic dependency

BUG=webrtc:6828
NOTRY=True
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2717113002
Cr-Commit-Position: refs/heads/master@{#17116}
2017-03-08 13:42:26 +00:00
ec304f96b3 GetTransportFeedbackVector return vector with lost packets too, sorted by seq-num
1. GetTransportFeedbackVector will now return a vector which also explicitly states lost packets.
2. The returned vector is unsorted (uses default order - by sequence number). It's up to the users to sort otherwise, if they need a different order.

BUG=None

Review-Url: https://codereview.webrtc.org/2707383006
Cr-Commit-Position: refs/heads/master@{#17114}
2017-03-08 13:03:53 +00:00
727ac1d4c0 Enable GN check for webrtc/logging
BUG=webrtc:6828, webrtc:7257
NOTRY=True

Review-Url: https://codereview.webrtc.org/2717903002
Cr-Commit-Position: refs/heads/master@{#17111}
2017-03-08 10:12:11 +00:00
0d4e235a96 Revert of Temporarily disable failing video_quality_loopback_test.py (patchset #1 id:1 of https://codereview.webrtc.org/2704073002/ )
Reason for revert:
Enabling this again so we can start running it experimentally on the bots.

Original issue's description:
> Temporarily disable failing video_quality_loopback_test.py
>
> BUG=webrtc:7185
> TBR=mandermo@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2704073002 .
> Cr-Commit-Position: refs/heads/master@{#16697}
> Committed: 6951a28b41

TBR=mandermo@webrtc.org
BUG=webrtc:7185
NOTRY=True

Review-Url: https://codereview.webrtc.org/2734333002
Cr-Commit-Position: refs/heads/master@{#17110}
2017-03-08 10:10:21 +00:00
9900be313c Revert of Delete unused TaskRunner abstraction. (patchset #2 id:20001 of https://codereview.webrtc.org/2622923002/ )
Reason for revert:
I had missed updating a few of Chrome's #includes, breaking the build.

Original issue's description:
> Delete unused TaskRunner abstraction.
>
> This is the fifth and final step in the process started in cl https://codereview.webrtc.org/2696703009/
>
> Depends on the landing of a copy of this code in Chrome (step 4), cl
> https://codereview.chromium.org/2694903005/
>
> BUG=webrtc:6424
>
> Review-Url: https://codereview.webrtc.org/2622923002
> Cr-Commit-Position: refs/heads/master@{#17107}
> Committed: 2d15fdd91b

TBR=pthatcher@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2732363004
Cr-Commit-Position: refs/heads/master@{#17109}
2017-03-08 10:01:07 +00:00
8d73f8c6fa Remove VoEVolumeControl interface.
BUG=webrtc:4690, webrtc:6206

Review-Url: https://codereview.webrtc.org/2727063004
Cr-Commit-Position: refs/heads/master@{#17108}
2017-03-08 09:52:20 +00:00
2d15fdd91b Delete unused TaskRunner abstraction.
This is the fifth and final step in the process started in cl https://codereview.webrtc.org/2696703009/

Depends on the landing of a copy of this code in Chrome (step 4), cl
https://codereview.chromium.org/2694903005/

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2622923002
Cr-Commit-Position: refs/heads/master@{#17107}
2017-03-08 09:45:05 +00:00
34b7a91862 Enable GN check for webrtc/modules/video_processing
BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2719753002
Cr-Commit-Position: refs/heads/master@{#17106}
2017-03-08 09:41:14 +00:00
3ae7c259ef Disable failing 15 thumbernails full-stack test on windows
BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/2737703005
Cr-Commit-Position: refs/heads/master@{#17105}
2017-03-08 09:17:35 +00:00
9fa7e4a0ac Fix error handling in X11 screen capturer
Previusly errors from XServerPixelBuffer::CaptureRect() were not always
handled, which results in a black frame returned from the capturer
instead of an error.

BUG=webrtc:7305

Review-Url: https://codereview.webrtc.org/2738513005
Cr-Commit-Position: refs/heads/master@{#17101}
2017-03-08 01:02:20 +00:00
b09b3f9a62 Add the option to disable IPv6 ICE candidates on WiFi.
Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.

BUG=b/35725283

Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
2017-03-07 22:40:51 +00:00
cb9ba301f0 Perform probing on network route change.
BUG=webrtc:7208

Review-Url: https://codereview.webrtc.org/2714503002
Cr-Commit-Position: refs/heads/master@{#17096}
2017-03-07 14:30:59 +00:00
f89a738626 Disable failing fullstack test with 15 thumbnail streams
BUG=webrtc:7301

Review-Url: https://codereview.webrtc.org/2739613003
Cr-Commit-Position: refs/heads/master@{#17095}
2017-03-07 14:15:27 +00:00
566c43b525 Cleaning up full-stack simulcast tests and making them more realistic.
BUG=none

Review-Url: https://codereview.webrtc.org/2734753003
Cr-Commit-Position: refs/heads/master@{#17093}
2017-03-07 12:42:54 +00:00
a014cc5eb1 Reland of "Added large room scenario to full-stack tests"
Added thumbnail streams functionality to video quality test.

Changed simulcast full-stack tests to be 30fps instead of 50 to
better reflect real usecases (expect all kind of perf metrics to
improve).

BUG=webrtc:7095, webrtc:7301

Review-Url: https://codereview.webrtc.org/2733943003
Cr-Commit-Position: refs/heads/master@{#17092}
2017-03-07 12:21:04 +00:00
d461ffce2a Avoid overflow in WebRtcSpl_DotProductWithScale
BUG=chromium:676935

Review-Url: https://codereview.webrtc.org/2717123004
Cr-Commit-Position: refs/heads/master@{#17091}
2017-03-07 12:02:47 +00:00
57f19cc0cd Drop VP8 frames in case of duplicates in RtpFrameReferenceFinder.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2734453002
Cr-Commit-Position: refs/heads/master@{#17090}
2017-03-07 11:54:05 +00:00
21dc1890f4 Replace Clock::CurrentNtp with Clock::CurrentNtpTime
CurrentNtp return time by taking two output parameters by reference
(also breaks style guide)
CurrentNtpTime treat ntp time as single entity and returns it using NtpTime structure.
(making interface clearer)

BUG=None

Review-Url: https://codereview.webrtc.org/2733823002
Cr-Commit-Position: refs/heads/master@{#17088}
2017-03-07 10:51:09 +00:00
92a7a1810c Update formatting of AudioLevel class
These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name.

BUG=none

Review-Url: https://codereview.webrtc.org/2731993002
Cr-Commit-Position: refs/heads/master@{#17087}
2017-03-07 09:58:55 +00:00
759358c9db Remove unused VoiceChannelTransport.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2687843002
Cr-Commit-Position: refs/heads/master@{#17086}
2017-03-07 09:47:57 +00:00
7cb0e55823 Remove voe_cmd_test.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2681993004
Cr-Commit-Position: refs/heads/master@{#17085}
2017-03-07 09:43:21 +00:00
17b958c041 Support pipelining codecs in VideoProcessor.
This CL removes most of the global frame state in VideoProcessor and
replaces that with a vector of frame states. This is useful for pipelining
codecs, where the encoded/decoded frame may not be immediately outputted
after it has been sent to the codec.

The callers (VideoProcessorIntegrationTest and video_quality_measurement)
still call VideoProcessor in a sequential fashion. A follow-up CL will be
submitted that enables batch mode in VideoProcessorIntegrationTest.

Note that VideoProcessor is still not thread safe. Currently, we can run
fairly well on Android due to the synchronicity of our MediaCodec wrapper,
but we still cannot run on iOS due to async issues. This will be fixed in
the future.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2711133002
Cr-Commit-Position: refs/heads/master@{#17084}
2017-03-07 09:41:43 +00:00
ae9ba047c4 Minor changes in videoprocessor and videoprocessor_integrationtests.h
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2708993005
Cr-Commit-Position: refs/heads/master@{#17083}
2017-03-07 08:25:38 +00:00
846e1be85c Fix iOS8 crash in background mode.
Add system version check functionality in UIDevice+RTCDevice category.
Check for iOS system version when handle capture session interruption.

BUG=webrtc:7201

Review-Url: https://codereview.webrtc.org/2733773003
Cr-Commit-Position: refs/heads/master@{#17079}
2017-03-07 00:42:19 +00:00
6dfd53a81e Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange
for consistency with the WebRTC 1.0 standard as suggested in a TODO.

BUG=None

Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
2017-03-06 21:49:03 +00:00
ad94c4c5d9 Replace StunMessage's vector<StunAttribute*>* with a
vector<unique_ptr<StunAttribute>> as suggested in a TODO.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2735523002
Cr-Commit-Position: refs/heads/master@{#17076}
2017-03-06 21:36:05 +00:00
d5a2d9ad0c WebRtcVideoChannel2Test::SetRecvCodecsAcceptsMultipleVideoCodecs passes now.
WebRtcVideoChannel2Test::SetRecvCodecsSetsFecForAllVideoCodecs was never
fully implemented and hasn't been touched in over a year.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2736483002
Cr-Commit-Position: refs/heads/master@{#17075}
2017-03-06 20:09:24 +00:00
eaa9c1db73 Remove HAVE_SRTP define and unmaintained code.
It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained
and failed to compile.

BUG=webrtc:7294

Review-Url: https://codereview.webrtc.org/2729373002
Cr-Commit-Position: refs/heads/master@{#17074}
2017-03-06 19:32:22 +00:00
bfb124596e Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
Reason for revert:
webrtc_perf_tests crashes on android and windows due to too large test.

Original issue's description:
> Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
>
> Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).
>
> BUG=webrtc:7095
>
> Review-Url: https://codereview.webrtc.org/2730073002
> Cr-Commit-Position: refs/heads/master@{#17068}
> Committed: d8bd1b1d82

TBR=sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2734753004
Cr-Commit-Position: refs/heads/master@{#17071}
2017-03-06 15:35:13 +00:00
b072739f8b Check __GLIBC_PREREQ availability before use.
BUG=webrtc:7287

Review-Url: https://codereview.webrtc.org/2727783004
Cr-Commit-Position: refs/heads/master@{#17070}
2017-03-06 15:34:06 +00:00
e7811f5d69 Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate.
BUG=webrtc:7105

Review-Url: https://codereview.webrtc.org/2733803002
Cr-Commit-Position: refs/heads/master@{#17069}
2017-03-06 14:49:27 +00:00
d8bd1b1d82 Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
Changed simulcast full-stack tests to be 30fps instead of 50 to better reflect real usecases (expect all kind of perf metrics to improve).

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2730073002
Cr-Commit-Position: refs/heads/master@{#17068}
2017-03-06 14:10:28 +00:00
a2ef4f94e4 Enable GN check for webrtc/media
BUG=webrtc:6828, webrtc:7245
NOTRY=True

Review-Url: https://codereview.webrtc.org/2716143002
Cr-Commit-Position: refs/heads/master@{#17067}
2017-03-06 14:04:55 +00:00
f949000834 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
BUG=None

Review-Url: https://codereview.webrtc.org/2710093004
Cr-Commit-Position: refs/heads/master@{#17066}
2017-03-06 13:32:21 +00:00
d1587ad244 Android HW decoder: Don't check slice_height for texture output
The check: 'RTC_CHECK_GE(slice_height, height);' has been observed to
fail after a reconfig. It looks like |slice_height| is still using the
previous resolution. |slice_height| isn't used for texture output and
hopefully this issue is texture specific. This CL only extracts and
checks |slice_height| when it's actually used.

BUG=b/35932686

Review-Url: https://codereview.webrtc.org/2736603003
Cr-Commit-Position: refs/heads/master@{#17065}
2017-03-06 13:20:49 +00:00
8f8d1a06b9 Adding placeholder for low bandwidth audio test
This will allow the trybots to be updated to start running this new test
executable, so that they can be used when landing this CL which will
replace the dummy test with real code:
https://codereview.webrtc.org/2694203002

Most likely, the trybots will just run the test binary while the perf bots
will run a Python wrapper script that takes care of the post-processing
to calculate audio quality using PESQ.

BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2717683002
Cr-Commit-Position: refs/heads/master@{#17063}
2017-03-06 12:01:16 +00:00
c898825eb4 Add kjellander to OWNERS for *.py in examples/android{app,tests}
BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2735673002
Cr-Commit-Position: refs/heads/master@{#17062}
2017-03-06 11:56:57 +00:00
b8102e0634 Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ )
Reason for revert:
The issue is now hopefully fixed.

Original issue's description:
> Revert of Add QP for FFmpeg H264 decoder. (patchset #4 id:200001 of https://codereview.webrtc.org/2649133007/ )
>
> Reason for revert:
> Let's revert this while we investigate a problem in H264 bitstream parser.
>
> Original issue's description:
> > Add QP for FFmpeg H264 decoder.
> >
> > BUG=webrtc:6541
> >
> > Review-Url: https://codereview.webrtc.org/2649133007
> > Cr-Commit-Position: refs/heads/master@{#16942}
> > Committed: 879f4f6c31
>
> TBR=sprang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6541, chromium:697795
>
> Review-Url: https://codereview.webrtc.org/2726973003
> Cr-Commit-Position: refs/heads/master@{#16974}
> Committed: 4c6df8893e

TBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6541, chromium:697795

Review-Url: https://codereview.webrtc.org/2735733002
Cr-Commit-Position: refs/heads/master@{#17061}
2017-03-06 11:52:55 +00:00
fb1fa44d70 Remove MockRemoteBitrateObserver (unused)
BUG=None

Review-Url: https://codereview.webrtc.org/2731523002
Cr-Commit-Position: refs/heads/master@{#17060}
2017-03-06 11:48:14 +00:00
609ab2b3dc Make ExtendedReports::target_bitrate() accessor const
as it supposed to be

BUG=None

Review-Url: https://codereview.webrtc.org/2726843007
Cr-Commit-Position: refs/heads/master@{#17059}
2017-03-06 11:12:12 +00:00
5419ac8c02 Remove unused RemoteBitrateEstimator::IncomingPacketFeedbackVector()
BUG=None

Review-Url: https://codereview.webrtc.org/2721463003
Cr-Commit-Position: refs/heads/master@{#17058}
2017-03-06 11:11:06 +00:00
dea7f4f46f Ignore aud and sei NALus when parsing bitstream.
We currently don't know how to parse these NALus and we don't need
any information from them anyway so we might as well skip parsing them
and not break.

BUG=chromium:697795

Review-Url: https://codereview.webrtc.org/2732623002
Cr-Commit-Position: refs/heads/master@{#17057}
2017-03-06 10:49:36 +00:00
5571ee9465 Fix memory leaks in Windows core audio
BUG=webrtc:7270

Review-Url: https://codereview.webrtc.org/2727273002
Cr-Commit-Position: refs/heads/master@{#17056}
2017-03-06 10:24:42 +00:00
3b2fb203fd Add PESQ precompiled tool for audio quality testing
BUG=webrtc:7229

Review-Url: https://codereview.webrtc.org/2715933003
Cr-Commit-Position: refs/heads/master@{#17055}
2017-03-06 10:23:34 +00:00
1993b1de1f Reland "Enable GN check for webrtc/examples"
This is a reland of https://codereview.webrtc.org/2714343002
with the errors related to inclusions of test targets in webrtc/api
resolved.

BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2733673002
Cr-Commit-Position: refs/heads/master@{#17053}
2017-03-06 08:29:21 +00:00
dfcab728b2 Reland: Improve testing of SRTP external auth code paths.
This CL is a reland of https://codereview.webrtc.org/2722423003 which got
reverted due to compile errors when rolling into Chromium.

Original CL description:

Improve testing of SRTP external auth code paths.

Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)

This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.

BUG=chromium:628400

Review-Url: https://codereview.webrtc.org/2735613002
Cr-Commit-Position: refs/heads/master@{#17052}
2017-03-06 08:14:10 +00:00