20238 Commits

Author SHA1 Message Date
Mirko Bonadei
8ed8e56149 Adding RTC_ prefixed LOG macros.
In order to avoid conflicts with downstream projects WebRTC is going
to prefix its LOG macros with RTC_.

This CL renames all the LOG macros to macros with the RTC_ prefix and
it also defines backward compatibility LOG macros in order to let
downstream projects to switch to RTC_ prefixed macros without breaking
them.

A follow-up CL will remove the usage of LOG macros in WebRTC.

Bug: webrtc:8452
Change-Id: Ic3e495cba6c772f65259dc65ee278560a59d02d7
Reviewed-on: https://webrtc-review.googlesource.com/15442
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20456}
2017-10-27 08:24:37 +00:00
Zhi Huang
b2d355ed1f Reland: Reject the description with fewer m= sections.
If the subsequent offer contains fewer m= sections than the existing
description, it would be rejected.

The helper method MediaSectionsInSameOrder is modified and it will
compare the number of m= sections before matching the media type.

The original CL: https://webrtc-review.googlesource.com/c/src/+/9621
Reland it after the web-platform-tests are updated:
https://chromium-review.googlesource.com/c/chromium/src/+/736318

TBR=deadbeef@webrtc.org

Bug: chromium:773620
Change-Id: I60e972eb856efc3cef4a18777791053c9f8e0491
Reviewed-on: https://webrtc-review.googlesource.com/16382
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20455}
2017-10-27 01:07:27 +00:00
Steve Anton
36b28db887 Fix clang style warnings in api/candidate.h
Bug: webrtc:163
Change-Id: I694194d34573b1f2e4769bf52ec861127940d017
Reviewed-on: https://webrtc-review.googlesource.com/15940
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20454}
2017-10-26 23:22:18 +00:00
Steve Anton
9d4a2e6fd2 Move inlined methods from p2p/base/packetsocketfactory.h
Bug: webrtc:163
Change-Id: I9c1eee77032326c2cafc38dabdd415583f9e1817
Reviewed-on: https://webrtc-review.googlesource.com/16067
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20453}
2017-10-26 23:14:37 +00:00
Zhi Huang
0af34ad3fa The onWebRtcAudioTrackStartError is changed in this CL which breaks the internal projects.
Revert "Improves native Android audio implementations."

This reverts commit 92b1ffd0f655e88532cb7313707f300fec911b46.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Improves native Android audio implementations.
> 
> Summary:
> 
> Adds AudioTrackStartErrorCode to separate different types of error
> codes in combination with StartPlayout.
> 
> Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
> to ensure that init/start/stop is performed identically.
> 
> Adds thread checking in WebRtcAudio track.
> 
> Bug: webrtc:8453
> Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
> Reviewed-on: https://webrtc-review.googlesource.com/15481
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20448}

TBR=henrika@webrtc.org,glaznev@webrtc.org

Change-Id: If1d1d9717387a4a8f6d9d6acf7e86ded4c655b5e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8453
Reviewed-on: https://webrtc-review.googlesource.com/16321
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20452}
2017-10-26 21:58:39 +00:00
Zijie He
c2a0eb2699 [Window Capture] Mouse cursor missing during window sharing on Mac OSX
CGWindowID is 32-bit, WindowId is 64-bit, using WindowId to receive int value
from CFNumberGetValue() causes the top 32 bits to be random. WindowFinderMac is
impacted by this issue and returns a random number. WindowCapturerMac cannot
match the window_id_ with the the random number.

Meanwhile MouseCursorMonitorMac uses window title to match "Dock" window. See,
https://cs.chromium.org/chromium/src/third_party/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm?rcl=a194e58e799ccab6c999998e5d0f75725aa3f748&l=174

This logic should not be necessary on 10.12 or upper, the name of dock window
is not "Dock" anymore. But to ensure the consistency on old platforms, I have
also added this logic back into GetWindowList() function.

Bug: chromium:778049
Change-Id: Ie827bcd5d31f2ca69ff24c24cf640cb7cc50d419
Reviewed-on: https://webrtc-review.googlesource.com/15782
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20451}
2017-10-26 21:14:57 +00:00
Sami Kalliomäki
9828bebee6 Fix/suppress all javac warnings.
This is done in preparation to make all javac warnings into errors for
WebRTC targets.

Bug: webrtc:6597
Change-Id: I402043157bd75943adf0de52111e5a1bb179c6d1
Reviewed-on: https://webrtc-review.googlesource.com/15104
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20450}
2017-10-26 15:11:36 +00:00
Sami Kalliomäki
ef5df1ae52 Fix WebSocketObserver getting garbage collected.
Apparently WebSocketObserver gets garbage collected if it is not stored
by us. This caused some external tests to break.

Bug: None
Change-Id: If62786e84f84a5a63172d67962bb4de8ae3e8479
Reviewed-on: https://webrtc-review.googlesource.com/16100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20449}
2017-10-26 14:24:06 +00:00
henrika
92b1ffd0f6 Improves native Android audio implementations.
Summary:

Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.

Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.

Adds thread checking in WebRtcAudio track.

Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
2017-10-26 13:45:36 +00:00
Danil Chapovalov
78161ca59d Add sending sdes to RtcpTransceiver.
Bug: webrtc:8239
Change-Id: Icff1528e177e0bb39dd82bd4f8533e1ed2736c40
Reviewed-on: https://webrtc-review.googlesource.com/15540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20447}
2017-10-26 13:33:16 +00:00
Oskar Sundbom
4556f3c3f4 Revert "Simple Default ObjC video codec factories."
This reverts commit 30915a742d86df55ac5c04501c0e8104675a612e.

Reason for revert: Breaks downstream.

Original change's description:
> Simple Default ObjC video codec factories.
> 
> Move the simple video encoder/decoder factory from AppRTCMobile into the
> public API so users who don't have special requirements for video codecs
> can easily get started.
> 
> Also clean up the API a little.
> 
> This CL replaces the more flexible default factories in
> https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
> want to implement their own codecs will have to supply their own
> encoder/decoder factories as well. The benefits of the approach in
> this CL are a simpler API and less effects on the rest of the code.
> 
> Bug: None
> Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
> Reviewed-on: https://webrtc-review.googlesource.com/15141
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20441}

TBR=magjed@webrtc.org,andersc@webrtc.org,kthelgason@webrtc.org

Change-Id: I3d4395cc9667e6c6cdb33a3b0f5c5fb5bfde9028
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/15182
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20446}
2017-10-26 11:55:57 +00:00
Mirko Bonadei
c15b2ddda9 Switch metrics_default and field_trial_default to source_set.
Bug: None
Change-Id: Ice987cca1f8720db0a5a1053977e1f463fada1c6
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/15323
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20445}
2017-10-26 11:47:46 +00:00
Magnus Jedvert
ef20795a46 Remove FlexFEC feedback params from internal encoder factory
I want to move away from the old encoder factory interface
cricket::WebRtcEncoderFactory to the new webrtc::VideoEncoderFactory. I
created a new webrtc::SdpVideoFormat that essentially is a subset of the
cricket::VideoCodec variables. E.g. the encoder factories shouldn't have
to assign payload types to the codecs, so the payload is not part of
webrtc::SdpVideoFormat. I also didn't add the "feedback_params" that is
used in cricket::VideoCodec to webrtc::SdpVideoFormat. This is causing
problems now, because the internal encoder factory is adding flexfec
feedback params. To avoid this problem, I add these feedback params in
WebRtcVideoEngine instead, like we do for the other codecs.

Bug: webrtc:7925
Change-Id: I7c6ae8d1e1f47f3631c4804c223ec21da8d73685
Reviewed-on: https://webrtc-review.googlesource.com/15223
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20444}
2017-10-26 11:33:27 +00:00
Christoffer Rodbro
d2817d80b5 Allow injection of custom network models in place of FakeNetworkPipe.
Adds a constructor for DirectTransport that takes a pointer to an instance 
of a class derived from FakeNetworkPipe. Said class can override Process() 
and SendPacket(...) members thereby emulating any desired network behavior.

Bug: b/67487983
Change-Id: I829fd3506124db61587af19192a14fdf62b06ca5
Reviewed-on: https://webrtc-review.googlesource.com/14620
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20443}
2017-10-26 11:11:25 +00:00
Bjorn Terelius
0295a967c0 Estimate RTP clock frequency and plot capture-send delay.
Bug: webrtc:8450
Change-Id: Idea093854a644f3018a565168425583dc4783ce9
Reviewed-on: https://webrtc-review.googlesource.com/15480
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20442}
2017-10-26 08:42:54 +00:00
Anders Carlsson
30915a742d Simple Default ObjC video codec factories.
Move the simple video encoder/decoder factory from AppRTCMobile into the
public API so users who don't have special requirements for video codecs
can easily get started.

Also clean up the API a little.

This CL replaces the more flexible default factories in
https://webrtc-review.googlesource.com/c/src/+/7741 and clients that
want to implement their own codecs will have to supply their own
encoder/decoder factories as well. The benefits of the approach in
this CL are a simpler API and less effects on the rest of the code.

Bug: None
Change-Id: I4ed94090d778b4fc38b49864de1d4de4ff125d6a
Reviewed-on: https://webrtc-review.googlesource.com/15141
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20441}
2017-10-26 08:31:54 +00:00
Niels Möller
a0565999db Delete VCMSendStatisticsCallback and corresponding use of ProcessThread
Bug: webrtc:8422
Change-Id: I5863266a0226d475c4fdd810f2f6f1acdf922df3
Reviewed-on: https://webrtc-review.googlesource.com/14880
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20440}
2017-10-26 08:13:55 +00:00
Sami Kalliomäki
2729c16143 Fix some Android lint warnings in AppRTCMobile.
Bug: webrtc:6597
Change-Id: I73e304ff03a5fcb166ff7bca61319904ef495426
Reviewed-on: https://webrtc-review.googlesource.com/15322
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20439}
2017-10-26 07:34:54 +00:00
Sami Kalliomäki
68e56a5951 Android: Update VideoDecoderFactoryWrapper to implement CreateVideoDecoderWithParams.
Old CreateVideoDecoder interface is deprecated. This allows
VideoDecoderFactoryWrapper to create codecs for types that WebRTC
doesn't know about.

Bug: webrtc:8140
Change-Id: I69aa1a0164642b4e4377daa1abeb9039c04fd884
Reviewed-on: https://webrtc-review.googlesource.com/15401
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20438}
2017-10-26 07:32:34 +00:00
Rasmus Brandt
fb6d32602c Delete unused PredictivePacketManipulator.
BUG=webrtc:8448

Change-Id: I07ff9db5cb49f84d98b6076e748a990aa560b5b5
Reviewed-on: https://webrtc-review.googlesource.com/15400
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20437}
2017-10-26 07:17:24 +00:00
Jerome Jiang
f7eea2a6bc Change key frame mismatch threshold for VP9 in unit test.
Change the threshold in ProcessNoLossChangeBitRateVP9.

Bug: webrtc:8442
Change-Id: Ic924a60f60c57cc2c990430cb6c70fdbefec97f4
Reviewed-on: https://webrtc-review.googlesource.com/15840
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20436}
2017-10-26 07:01:15 +00:00
Steve Anton
eae3e65394 Enable the clang style plugin for the stunprober target
Bug: webrtc:163
Change-Id: I77c92dfc05626cb5e83d4e93d735c1370ed4af23
Reviewed-on: https://webrtc-review.googlesource.com/15783
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20435}
2017-10-26 04:40:25 +00:00
Steve Anton
ca7d54e16a Fix clang style warnings in p2p/base/stun.h
Bug: webrtc:163
Change-Id: Ief9c59f80f36d3339fd40bed9f33e8c6eeef4f90
Reviewed-on: https://webrtc-review.googlesource.com/15781
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20434}
2017-10-26 04:36:34 +00:00
Per Åhgren
74e72c8c9b Lowering the threshold for delay change detection in AEC3
This CL lowers the threshold for delay change detection in AEC3.
This makes the delay decisions more stable.

TBR=gustaf@webrtc.org

Bug: chromium:778396,webrtc:8451
Change-Id: I8b015455399d696172b7c0beb033caf508f426e9
Reviewed-on: https://webrtc-review.googlesource.com/15541
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20433}
2017-10-25 21:56:30 +00:00
Zijie He
173fd91b56 [Window Capture] Inaccurate cursor position during window sharing on X11
{root_x, root_y} should be used to report the absolute cursor position in
MouseCursorMonitorX11.

Bug: chromium:778035
Change-Id: I421005d52786a57da8e8c3901bdf4afa2843ff24
Reviewed-on: https://webrtc-review.googlesource.com/15680
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20432}
2017-10-25 20:41:29 +00:00
Gustaf Ullberg
84634b8634 Temporarily disabled failing death test.
Some death tests for AEC3 cause memory leaks on trybots. This CL
temporarily disables BlockProcessor.VerifyRenderBlockSizeCheck.

Bug: webrtc:8449,webrtc:6985
Change-Id: I2900a73f7c7d5bf0e8b58a20f9a40bd5d654629a
Reviewed-on: https://webrtc-review.googlesource.com/15500
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20431}
2017-10-25 15:24:46 +00:00
Alex Loiko
b9f536167c Removing undefined left shifts in AudioProcessing
This CL replaces 5 left shifts where the shifted value may be 
negative. The shifts are replaced with equivalent multiplications.

Bug: chromium:777231, chromium:776719, chromium:776624, chromium:776286
Change-Id: Ifb27d5506eac779e60f238432bdf9e4bc5b2da4c
Reviewed-on: https://webrtc-review.googlesource.com/14800
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20430}
2017-10-25 13:35:36 +00:00
Bjorn Terelius
a194e58e79 Move sequence_number_utils.h to rtc_base/
Bug: webrtc:8440
Change-Id: I36e70da6ce70b95db7d3fce8b0013bff5c795bfc
Reviewed-on: https://webrtc-review.googlesource.com/14860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20429}
2017-10-25 12:33:57 +00:00
Alex Loiko
ddfd9c5fd2 Fix AudioProcessing fuzzer crash.
When audio_processing_fuzzer runs with 'DCHECK_ALWAYS_ON', it crashes
when both AEC and AECM is enabled at the same time. This change
detects that case and fixes
https://clusterfuzz.com/v2/testcase-detail/6389429496446976.

It also removes an unnecessary safeguard that didn't allow fuzzing
with 8kHz input signals.

Bug: chromium:776358
Change-Id: I33c18a2a235e50ae410f7be24637872823e432eb
Reviewed-on: https://webrtc-review.googlesource.com/15320
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20428}
2017-10-25 12:00:36 +00:00
Karl Wiberg
ef52d8b859 Presubmit: Don't forget to warn when changing headers in subdirs of api/
Unlike all the other API directories, api/ is the root of an entire
tree of directories that are also API directories.

BUG=webrtc:8445

Change-Id: I218befe6fb6113b95599512f062ebe63abc98889
Reviewed-on: https://webrtc-review.googlesource.com/15321
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20427}
2017-10-25 11:53:16 +00:00
Mirko Bonadei
d71997941a Adding win_more_configs to CQ
Bug: chromium:759980
Change-Id: Ie33931eae67b90a648735856a26e3b86dcf7c0e1
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14960
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20426}
2017-10-25 11:51:36 +00:00
Kári Tristan Helgason
47d3a0197f Reenable some supressed warnings for the objc SDK.
Bug: webrtc:8441
Change-Id: I6b427dfc1fe275e274d042766e0850628cf19994
Reviewed-on: https://webrtc-review.googlesource.com/15000
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20425}
2017-10-25 11:17:36 +00:00
Karl Wiberg
7275e18439 Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix
We've done this previously with the other audio encoders, but Opus had
to wait until all external users had been updated.

BUG=webrtc:7847

Change-Id: I70422d7b6c715f32a43bee88febcf6b6155e18b3
Reviewed-on: https://webrtc-review.googlesource.com/8000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20424}
2017-10-25 10:19:06 +00:00
Erik Språng
7c8cca3dce Add check for send-side bwe before applying alr settings
Bug: webrtc:7694
Change-Id: I359b27b96239af4e067055fc77ea285824e69edf
Reviewed-on: https://webrtc-review.googlesource.com/14603
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20423}
2017-10-25 09:55:06 +00:00
Rasmus Brandt
58b72914d8 Log warning when receiving an H.264 containing IDR, but not SPS/PPS.
BUG=webrtc:8423

Change-Id: Ica8cb5062b9b8b4b7f2c0e569a5ce5d2dc9effc7
Reviewed-on: https://webrtc-review.googlesource.com/15220
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20422}
2017-10-25 09:45:06 +00:00
Ilya Nikolaevskiy
d79314f9f9 Reland "Add fine grained dropped video frames counters on sending side"
Add fine grained dropped video frames counters on sending side

4 new counters added to SendStatisticsProxy and reported to UMA and logs.

Bug: webrtc:8355
Change-Id: I1f9bdfea9cbf17cf38b3cb2f55d406ffdb06614f
Reviewed-on: https://webrtc-review.googlesource.com/14580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20421}
2017-10-25 09:32:15 +00:00
Åsa Persson
f0c44672df Make VideoProcessor::Init/Release methods private and call from constructor/destructor.
TestConfig: Replace Print method with ToString and add test.

Bug: none
Change-Id: I9853cb16875199a51c5731d1cec326159751d001
Reviewed-on: https://webrtc-review.googlesource.com/14320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20420}
2017-10-25 09:31:12 +00:00
Ilya Nikolaevskiy
c22a3a6a7d Refactor VP8 encoder creation logic
Now decision between using SimulcastEncoderAdapter and using VP8 encoder
is postponed before codec is initialized for VP8 internal codecs. This is done
be new VP8EncoderProxy class. New error code for codec initialization is used
to signal that simulcast parameters are not supported.

Bug: webrtc:7925
Change-Id: I3a82c21bf5dfaaa7fa25350986830523f02c39d8
Reviewed-on: https://webrtc-review.googlesource.com/13980
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20419}
2017-10-25 09:30:07 +00:00
Per Åhgren
7ddd46386a Balancing the transparency in AEC3 between saturating and low echo paths
This CL balances the NLP tradeoff in AEC3 to properly handle the cases
when the echo path is so strong that it saturates the echo and when it
is so weak that the echo is very low compared to nearend.

Bug: webrtc:8411, webrtc:8412, chromium:775653
Change-Id: I5aff74dfadd51cac1ce71b1cb935d68a5be6918d
Reviewed-on: https://webrtc-review.googlesource.com/14120
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20418}
2017-10-25 01:36:59 +00:00
Yuwei Huang
d9f99c1e7a Replace Atomic32 with std::atomic in video/
system_wrapper/Atomic32 has been deprecated (which is already just a
wrapper of std::atomic) in favor of platform-independent std::atomic
from C++11. This CL replaces all use of Atomic32 in video/

Bug: webrtc:8428
Change-Id: If4dab4909df06944c009e7b70141f58daef7be10
Reviewed-on: https://webrtc-review.googlesource.com/14720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yuwei Huang <yuweih@google.com>
Cr-Commit-Position: refs/heads/master@{#20417}
2017-10-24 23:40:29 +00:00
Steve Anton
9de3aaccc9 Reland "Enable the clang style plugin in rtc_base/"
This is a reland of Id63f0deb7b335690157ab157c35177b7836688da.

Original change's description:
> Enable the clang style plugin in rtc_base/
> 
> Enabled the plugin and cleaned up all issues it found.
> 
> Bug: webrtc:163
> Change-Id: Id63f0deb7b335690157ab157c35177b7836688da
> Reviewed-on: https://webrtc-review.googlesource.com/14660
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20401}

Bug: webrtc:163
Change-Id: I861a5fe741215115b0e7a2be9c0786836ff5376e
Reviewed-on: https://webrtc-review.googlesource.com/15040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20416}
2017-10-24 20:50:13 +00:00
Steve Anton
074dece085 Fix flaky DataChannel integration test
The DataChannel OPEN message is sent in-band in this test, and waiting
for the signaling state to change is not sufficient to guarantee that the
callee received this message

Bug: webrtc:8443
Change-Id: I76fa6348b6f8e1e70fb41a4e644aee805b2ef4de
Reviewed-on: https://webrtc-review.googlesource.com/15060
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20415}
2017-10-24 20:41:48 +00:00
Danil Chapovalov
398a7c67b1 Create skeleton of the rtcp transceiver.
RtcpTransceiver name reserved for thread-safe version that planned to
be wrapper of the RtcpTransceiverImpl

BUG=webrtc:8239

Change-Id: If8a3092eb1b8e4175e3efd23b52e1043cdabf19f
Reviewed-on: https://webrtc-review.googlesource.com/7920
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20414}
2017-10-24 19:35:38 +00:00
Steve Anton
d5585ca956 Move almost all references from WebRtcSession to PeerConnection
WebRtcSession is being merged into PeerConnection, and to make the
code review easier this is the first step towards achieving that.

Bug: webrtc:8323
Change-Id: I33778e46f20cb14089dff4328947868e207476bd
Reviewed-on: https://webrtc-review.googlesource.com/8760
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20413}
2017-10-24 17:59:20 +00:00
Steve Anton
c4faa9c4e1 Remove QUIC transport/data channel
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.

Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
2017-10-24 16:14:18 +00:00
Ilya Nikolaevskiy
a0b66c7566 Enable cpu adaptation for screenshare in chrome as field trial
Bug: webrtc:8433
Change-Id: I1632cb0191b06e99d090a090bad9db20d7a81349
Reviewed-on: https://webrtc-review.googlesource.com/14980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20411}
2017-10-24 15:07:38 +00:00
Dino Radaković
21360eb01e Add application extension field to RtpPacketReceived.
Bug: webrtc:8439
Change-Id: I372e90c81a68351d343554fb77ce6ef77d538e62
Reviewed-on: https://webrtc-review.googlesource.com/14820
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20410}
2017-10-24 14:22:18 +00:00
Rasmus Brandt
f7a3558f3e Add VideoProcessor tests verifying that H.264 keyframes contain SPS/PPS/IDR.
This CL adds an EncodedFrameChecker interface which can be used by users
of the VideoProcessor to inject customized per-frame checks to the
encoding/decoding pipeline. This currently has two uses:
- Verifying that the QP parser works correctly for VP8 and VP9, by comparing the
  parsed QP to that produced by libvpx.
- Verifying that our H.264 encoders always produce SPS/PPS/IDR in tandem.

TESTED=Galaxy S8, Pixel 2 XL, iPhone 7.
BUG=webrtc:8423

Change-Id: Ic3e401546e239a9ffaf2ed2907689cebb1127805
Reviewed-on: https://webrtc-review.googlesource.com/14559
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20409}
2017-10-24 13:58:38 +00:00
Rasmus Brandt
edf4ff7e0f Only treat H.264 frames containing SPS, PPS, and IDR as key frames.
This is protected behind a field trial, for controlled rollout.

TESTED=MediaCodec (Qualcomm + Exynos) and VideoToolbox senders.
BUG=webrtc:8423

Change-Id: Ibccefb3d374e4a44461d33e77eff754d8d752666
Reviewed-on: https://webrtc-review.googlesource.com/13863
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20408}
2017-10-24 11:51:18 +00:00
Bjorn Terelius
6984ad212c Print state of AcknowledgedBitrateEstimator in event_log_visualizer.
Bug: None
Change-Id: Iabf53be419ba94874619f417131674692172f6ba
Reviewed-on: https://webrtc-review.googlesource.com/14322
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20407}
2017-10-24 11:19:48 +00:00