22430 Commits

Author SHA1 Message Date
Autoroller
7d86b1fa7b Roll chromium_revision 4116d5a412..9d7b66e9bb (557414:557514)
Change log: 4116d5a412..9d7b66e9bb
Full diff: 4116d5a412..9d7b66e9bb

Changed dependencies:
* src/base: 98837a25c9..0c405c7590
* src/ios: dd6e85d5da..2790cdbe0f
* src/testing: 3eaaac8ed2..90eda7fc6f
* src/third_party: 80a527d468..c0a852a881
* src/third_party/depot_tools: b61d387fa2..7c3ff1311a
* src/tools: 5d0a6695d8..56a7c67683
DEPS diff: 4116d5a412..9d7b66e9bb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ib490fd1cd7e857ca2f0d189aa73a8f56d196d03f
Reviewed-on: https://webrtc-review.googlesource.com/75904
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23198}
2018-05-10 15:22:49 +00:00
Harald Alvestrand
7f0a069550 Reduce log level for socket.SetOptions() to LS_INFO
Bug: webrtc:9221
Change-Id: I7bbbece754afa4e02ab000ee33e2b09ead5647a1
Reviewed-on: https://webrtc-review.googlesource.com/73686
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23197}
2018-05-10 06:40:41 +00:00
Autoroller
b4e0e50a2b Roll chromium_revision d66554b28a..4116d5a412 (557284:557414)
Change log: d66554b28a..4116d5a412
Full diff: d66554b28a..4116d5a412

Changed dependencies:
* src/build: 8bdd49e0f7..1bdeddd932
* src/ios: 333d6aa044..dd6e85d5da
* src/testing: ad35d9fc8c..3eaaac8ed2
* src/third_party: 1b5595d935..80a527d468
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/483449aa72..4a13b64861
* src/third_party/depot_tools: afec759dca..b61d387fa2
* src/third_party/gtest-parallel: a8f5453ffc..cb3514a085
* src/tools: d85892881d..5d0a6695d8
DEPS diff: d66554b28a..4116d5a412/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I818ef5f55443292b627389a4264c24596f9d8f96
Reviewed-on: https://webrtc-review.googlesource.com/75842
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23196}
2018-05-10 02:20:21 +00:00
Qingsi Wang
e53ac0463d Add Ethernet and loopback entries to GetAdapterTypeFromName.
GetAdapterTypeFromName determines the adapter type of a network
interface based on the string matching of the interface name. It however
does not have an entry to map the well-known "eth" name to the common
Ethernet type. This introduces subtle bugs when GetAdapterTypeFromName
is used as the only method to determine a network type and Ethernet is
thus identified as an unknown network, which affects the network
filtering and network path selection that rely on the network type.

Bug: webrtc:9235
Change-Id: Ifc3269d191382f3b3a041de1c9755c09994b31b2
Reviewed-on: https://webrtc-review.googlesource.com/74263
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23195}
2018-05-10 00:18:11 +00:00
Autoroller
e1f222e5c5 Roll chromium_revision ac6b00ded1..d66554b28a (557117:557284)
Change log: ac6b00ded1..d66554b28a
Full diff: ac6b00ded1..d66554b28a

Changed dependencies:
* src/base: 51fc247a21..98837a25c9
* src/build: 7fe7b26db7..8bdd49e0f7
* src/ios: 2d0c659879..333d6aa044
* src/testing: afb6e381a5..ad35d9fc8c
* src/third_party: 671a6f40b1..1b5595d935
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/718dbe7d32..483449aa72
* src/third_party/depot_tools: 3806b7fbd0..afec759dca
* src/tools: b5ebf2fac7..d85892881d
DEPS diff: ac6b00ded1..d66554b28a/DEPS

Clang version changed 330570:331747
Details: ac6b00ded1..d66554b28a/tools/clang/scripts/update.py

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Iba0d7b7721f7060b4e5eefcdac0499393766e4ae
Reviewed-on: https://webrtc-review.googlesource.com/75701
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23194}
2018-05-09 20:30:41 +00:00
Sergey Silkin
377ef24a8f Remove extra reference from GOF.
This removes second reference for frame 3 in GOF predefined for 3
temporal layers since encoder never use that reference.

Bug: webrtc:9245
Change-Id: I6fbdbe7d3c753dda7fbcfcbd05f3530f70f80728
Reviewed-on: https://webrtc-review.googlesource.com/74705
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@google.com>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23193}
2018-05-09 18:18:43 +00:00
Henrik Lundin
a29b148557 Create a fuzzer for the Opus encoder
The fuzzer is very simple. It only considers the default encoder
configuration at this point.

Bug: chromium:826914
Change-Id: Ifa248a1dba80efb231807750e40082ec5580636a
Reviewed-on: https://webrtc-review.googlesource.com/75261
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23192}
2018-05-09 13:35:23 +00:00
Per Åhgren
d18e87edd4 Correcting the AEC3 transparent mode behavior avoid incorrect activation
This CL adds robustness to avoid the AEC3 transparent mode to be
incorrectly activated when
-there is strong near-end noise
-there is only low-level nearend activity.

Bug: webrtc:9256,chromium:841193
Change-Id: I26c2759d163914eb85dc3d863da8acbf28cbb88d
Reviewed-on: https://webrtc-review.googlesource.com/75511
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23191}
2018-05-09 12:36:41 +00:00
Per Åhgren
ced31ba1cf Correcting the usage of the estimated echo path gain in AEC3
This CL corrects the usage of the estimated echo path gain to not be
hardcoded to 1. In order to retain the tuned behavior, the CL for now
maintains the former behavior in the code.

Bug: webrtc:9255,chromium:851187
Change-Id: I7f91c72e476680a8a854c22b74b1771fae446110
Reviewed-on: https://webrtc-review.googlesource.com/75510
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23190}
2018-05-09 12:35:31 +00:00
Per Åhgren
e05c43cc39 Remove the headroom and delay estimation feedback loop in AEC3
This CL ensures that the external audio buffer delay is correctly used
by removing the applied headroom and avoiding that the delay estimation
feedback fromt the echo remover overrules the external delay
information.

Bug: webrtc:9241,chromium:839860
Change-Id: I53cc78ace34a71994ab24a3b552f29979e2aae78
Reviewed-on: https://webrtc-review.googlesource.com/75513
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23189}
2018-05-09 12:34:26 +00:00
Sebastian Jansson
c6c44268bc Moves network control interface to API.
This prepares for allowing injection of a network controller.

Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
2018-05-09 11:01:36 +00:00
Mirko Bonadei
7eca805ce3 Removing -Wno-unused-private-field.
This CL is part of the effort to remove warning suppression flags from
the WebRTC build.

Bug: webrtc:9251
Change-Id: I45ece25e897a14a6d4ce8a90ba59688f8fc6fe32
Reviewed-on: https://webrtc-review.googlesource.com/75503
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23187}
2018-05-09 10:55:56 +00:00
Mirko Bonadei
a05d47e344 Adding a way to disable public_deps presubmit check.
This is useful when someone is just moving code around or when there is
a good reason to use public_deps.

Example of the error message:
** Presubmit ERRORS **
public_deps is discouraged in WebRTC BUILD.gn files because it doesn't
map well to downstream build systems.
Used in: BUILD.gn (line 31).
If you are not adding this code (e.g. you are just moving existing code)
or you have a good reason, you can add a comment on the line that causes
the problem:

public_deps = [  # no-presubmit-check TODO(webrtc:8603)

Bug: webrtc:8603
Change-Id: If2645b6ba60c7cbf5416450cf6e5a8c08bf4934e
Reviewed-on: https://webrtc-review.googlesource.com/75508
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23186}
2018-05-09 10:18:06 +00:00
Mirko Bonadei
212fb5e4d8 Removing -Wno-tautological-compare.
Bug: webrtc:9251
Change-Id: I092fbb596dc67f7a381182e734d68709c730c5c0
Reviewed-on: https://webrtc-review.googlesource.com/75501
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23185}
2018-05-09 09:37:46 +00:00
Sami Kalliomäki
ee98be7811 Fix handling non-tightly packed ByteBuffers in HardwareVideoDecoder.
Before this CL, there would be an out-of-bounds write in the ByteBuffer
copying when a decoded frame had height != sliceHeight.

Bug: webrtc:9194
Change-Id: Ibb80e5555e8f00d9e1fd4cb8a73f5e4ccd5a0b81
Tested: 640x360 loopback with eglContext == null in AppRTCMobile on Pixel.
Reviewed-on: https://webrtc-review.googlesource.com/74120
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23184}
2018-05-09 09:15:46 +00:00
Mirko Bonadei
c710ac142e Removing -Wno-comment.
Chromium is suppressing this warning only on GCC [1], so WebRTC should
not suppress it on clang and just rely on Chromium's defaults.

[1] - https://cs.chromium.org/chromium/src/build/config/compiler/BUILD.gn?l=1356&rcl=027d7fa1c191f60f754985b9c235597f8c9a2081

Bug: webrtc:9251
Change-Id: I9316cbdda4083da7d859ff0b9c60579546ddbfcb
Reviewed-on: https://webrtc-review.googlesource.com/75301
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23183}
2018-05-09 09:12:06 +00:00
Autoroller
f4e8b9f511 Roll chromium_revision 3055ddf805..ac6b00ded1 (556795:557117)
Change log: 3055ddf805..ac6b00ded1
Full diff: 3055ddf805..ac6b00ded1

Changed dependencies:
* src/base: c5356a4b51..51fc247a21
* src/build: 83c3af53bd..7fe7b26db7
* src/ios: 4f5968682f..2d0c659879
* src/testing: 6953f9fb51..afb6e381a5
* src/third_party: 2578cbaf20..671a6f40b1
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/08f785343a..718dbe7d32
* src/third_party/depot_tools: 5ae86d2021..3806b7fbd0
* src/third_party/icu: e4194dc7bb..f61e46dbee
* src/tools: 0a04830d66..b5ebf2fac7
DEPS diff: 3055ddf805..ac6b00ded1/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Idc343514fae1c972dc44a4796ad227860d2c7d49
Reviewed-on: https://webrtc-review.googlesource.com/75485
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23182}
2018-05-09 08:38:06 +00:00
Minyue Li
5ebb416aaf Fixing NetEq RTP player.
A bug was introduced to NetEq RTP player in a recent CL:
https://webrtc-review.googlesource.com/c/src/+/69806

This is to fix it.

Bug: webrtc:9147
Change-Id: I949fd6b220d7c7f08c6e2940468232d1d955a3dc
Reviewed-on: https://webrtc-review.googlesource.com/75321
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23181}
2018-05-09 07:43:16 +00:00
Benjamin Wright
d6f86e8fca This changeset adds dependency injection support for SSL Root Certs.
This extends the API surface so that
custom certificates can be provided by an API user in both the standalone and
factory creation paths for the OpenSSLAdapter. Prior to this change the SSL
roots were hardcoded in a header file and directly included into
openssladapter.cc. This forces the 100 kilobytes of certificates to always be
compiled into the library. This is undesirable in certain linking cases where
these certificates can be shared from another binary that already has an
equivalent set of trusted roots hard coded into the binary.

Support for removing the hard coded SSL roots has also been added through a new
build flag. By default the hard coded SSL roots will be included and will be
used if no other trusted root certificates are provided.

The main goal of this CL is to reduce total binary size requirements of WebRTC
by about 100kb in certain applications where adding these certificates is
redundant.

Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f

Bug: chromium:526260
Change-Id: Ifd36d92b5cb32d1b3098a61ddfc244d76df8f30f
Reviewed-on: https://webrtc-review.googlesource.com/64841
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23180}
2018-05-09 00:24:05 +00:00
Seth Hampson
7c682e0c35 Update to allow the application to set a low max bitrate.
A bug surfaced when setting a low max bitrate with
30kbps hard-coded min bitrate value then a DCHECK was hit in the
VideoCodecInitializer, expecting the max bitrate to be higher than the
min bitrate. This change allows the application to set a max bitrate
below 30kbps, and adjusts the min bitrate to the value set for the
max bitrate.

RtpSender: :setParameters. If the value set was lower than the
Bug: webrtc:9141
Change-Id: I9b43ee7814b1a2caba00bc9614fc66d4438d66d8
Reviewed-on: https://webrtc-review.googlesource.com/74641
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23179}
2018-05-08 23:03:35 +00:00
Henrik Lundin
4e268edb53 Add two new RTP header extensions to neteq_rtpplay
This change adds flags and default values for two more RTP header
extensions: VideoContentType and VideoTiming.

This will silence a number of annoying warnings when running with
application logs.

Bug: none
Change-Id: I9bb01ea2519813d3c47553ecff384141fbede23e
Reviewed-on: https://webrtc-review.googlesource.com/75300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23178}
2018-05-08 16:05:12 +00:00
Sergey Silkin
acb4cba5b1 Ignore spatial index.
This workaround allows to decode VP9 SVC streams with partially enabled
inter-layer prediction.

This change won't affect conventional SVC (inter-layer prediction is
enabled for all frames) since spatial index was always zero in this
case.

Bug: webrtc:9249
Change-Id: If6ff26a18b7cf543ec9e7f70b9239e9edff250b5
Reviewed-on: https://webrtc-review.googlesource.com/74924
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23177}
2018-05-08 16:03:53 +00:00
Autoroller
1729711a70 Roll chromium_revision 5de2157f1e..3055ddf805 (556680:556795)
Change log: 5de2157f1e..3055ddf805
Full diff: 5de2157f1e..3055ddf805

Changed dependencies:
* src/build: b61b6b6a2e..83c3af53bd
* src/ios: 9d27efb09d..4f5968682f
* src/testing: f0ade05cb2..6953f9fb51
* src/third_party: 62736ae0ad..2578cbaf20
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e853531767..08f785343a
* src/tools: abafe60c0f..0a04830d66
DEPS diff: 5de2157f1e..3055ddf805/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ic0baa36b9e9a8d97ab581cb049320bb01e0c1483
Reviewed-on: https://webrtc-review.googlesource.com/75242
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23176}
2018-05-08 15:20:42 +00:00
Mirko Bonadei
03b41483c1 Removing warning suppression -Wno-missing-braces.
Bug: webrtc:9251
Change-Id: Ie32a052738d260364a7543e83e8b46ee3d34df59
Reviewed-on: https://webrtc-review.googlesource.com/75200
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23175}
2018-05-08 13:39:52 +00:00
Sebastian Jansson
5f83cf0c6d Replacing rtc::TimeDelta with webrtc::TimeDelta.
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.

Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
2018-05-08 13:22:53 +00:00
Åsa Persson
5b2b692079 Remove unused members in HistogramTest.
Bug: none
Change-Id: I594b8d03373703d0216fc85c51a16638649cf5f3
Reviewed-on: https://webrtc-review.googlesource.com/74580
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23173}
2018-05-08 12:55:02 +00:00
Paulina Hensman
24bebb86bd Add checks and offset when using byteBuffer in WebRtcAudioRecord.
See bug for more info.

In this case, the offset of the byteBuffer was observed to be 4 bytes
when testing, meaning that the first 4 bytes sent to the AudioSamples
callback were empty, and the last 4 bytes that should have been sent
were not sent.

This CL adjusts the range copied from the backing array to match the
offset.

Bug: webrtc:9175
Change-Id: I40ac6e10c6d7058ead7eff1c9fa2f342920cf2a4
Reviewed-on: https://webrtc-review.googlesource.com/75123
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23172}
2018-05-08 12:36:22 +00:00
Alex Loiko
0520b0eb7b FFT-based auto correlation.
During pitch search in the RNN VAD, we calculate auto
correlation. Before this CL, we computed kNumInvertedLags12kHz=147 dot
products of vectors with kBufSize12kHz-kMaxPitch12kHz=240
elements. This was the most time consuming step of the new VAD.

This CL makes the computation happen in frequency domain. Profiling
shows a 3x speed increase. In future, we can try using a more efficient
FFT and to reduce the FFT length to some of e.g. 400, 405, 432.

# For minimal Clang plugin check change.
TBR: kwiberg@webrtc.org

Bug: webrtc:9076
Change-Id: I688251a415869d53175a37f390f441d4e035d954
Reviewed-on: https://webrtc-review.googlesource.com/73366
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23171}
2018-05-08 12:07:42 +00:00
Alessio Bazzica
0bd0a3fe4c AGC2 RNN VAD: Spectral features internal API.
This CL adds helper functions to be used for the spectral features
computation. Namely, it includes the following:
- band boundaries (frequency to FFT coeffcient index)
- band energy coefficients
- log band energy coefficients
- fixed size DCT table and computation

Bug: webrtc:9076
Change-Id: I03a8799b226d986bc1e37cefd0c3039f94b5592a
Reviewed-on: https://webrtc-review.googlesource.com/73687
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23170}
2018-05-08 11:52:32 +00:00
Magnus Jedvert
496caa9095 Update sdk/objc ownership
Add new team members as owners of sdk/objc.

Bug: None
Change-Id: Id8c40fb018da2ab634bc1117afda555275a8b0f8
Reviewed-on: https://webrtc-review.googlesource.com/74002
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23169}
2018-05-08 11:48:32 +00:00
Sebastian Jansson
6fae6ec2ee Moves network unit types to API.
This prepares for being able to inject network congestion controllers.
And makes it easier to use the units in other parts of the code.

Bug: webrtc:9155
Change-Id: Ib8f9c1c97b06d791a01c3376046933d576ae46f9
Reviewed-on: https://webrtc-review.googlesource.com/70201
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23168}
2018-05-08 11:46:22 +00:00
Henrik Lundin
76c106725a ACM: Properly initialize last_audio_buffer_ array
Only half of the array was initialized. Now all of it is.

Bug: chromium:839960
Change-Id: If8bbe12c4c4c0dc0d529c93b22e49a94ecb09919
Reviewed-on: https://webrtc-review.googlesource.com/74820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23167}
2018-05-08 11:40:04 +00:00
Max Morin
826738b78c Revert "Moving iOS Audio Device to sdk."
This reverts commit a167212657f8450296ac518162ef0b28ba4214c5.

Reason for revert: Breaks Chromium build.
Log:
https://ci.chromium.org/buildbot/chromium.webrtc.fyi/ios-device/
Writing """\
additional_target_cpus = [ "arm64" ]
goma_dir = "/b/c/goma_client"
ios_enable_code_signing = false
is_component_build = false
is_debug = false
target_cpu = "arm"
target_os = "ios"
use_goma = true
""" to /b/c/b/ios_device/src/out/Release-iphoneos/args.gn.
/b/c/b/ios_device/src/buildtools/mac/gn gen //out/Release-iphoneos --check
  -> returned 1
ERROR at //third_party/webrtc/sdk/BUILD.gn:108:9: Can't load input file.
        "../../rtc_base:checks",
        ^----------------------
Unable to load:
  /b/c/b/ios_device/src/third_party/rtc_base/BUILD.gn
I also checked in the secondary tree for:
  /b/c/b/ios_device/src/build/secondary/third_party/rtc_base/BUILD.gn

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> Bug: webrtc:9120
> Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
> Reviewed-on: https://webrtc-review.googlesource.com/67300
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23163}

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

Change-Id: Iebe52e9775409a3bdd6d5e44f4f985d56b859cbe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/75220
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23166}
2018-05-08 11:00:37 +00:00
Rasmus Brandt
5f0d0e737e Re-enable PeerConnectionTest#testTrackRemovalAndAddition.
Let the test expect calls to onRenegotiationNeeded(), as introduced by
https://codereview.webrtc.org/2977493002.

Bug: webrtc:7761
Change-Id: If8e3c484236f6599cc225a0398bbbc9cf6c356a5
Reviewed-on: https://webrtc-review.googlesource.com/48364
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23165}
2018-05-08 09:57:26 +00:00
Alessio Bazzica
2284c56670 Adding double braces for array initialization.
TBR=maxmorin@webrtc.org

Bug: webrtc:9076
Change-Id: Ic341ef7437392dd5d6141147a2412ec54204ae10
Reviewed-on: https://webrtc-review.googlesource.com/75121
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23164}
2018-05-08 09:15:16 +00:00
Peter Hanspers
a167212657 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
2018-05-08 08:46:25 +00:00
Niels Möller
8df3a388a3 Deprecate RTPFragmentationHeader argument to VideoDecoder::Decode
Intend to delete in a later cl.

Bug: webrtc:6471
Change-Id: Icf0fcd40e0d3287dc59b684fae6552b40b47204a
Reviewed-on: https://webrtc-review.googlesource.com/39511
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23162}
2018-05-08 08:09:35 +00:00
Sebastian Jansson
ea562b40f8 Removes dependency on Optional for unit classes.
This makes the dependency graph simpler and prepares for moving the
unit classes to api/.

Bug: webrtc:9155
Change-Id: I1b36d5e05f75d70ba8951e880d76359f896f7741
Reviewed-on: https://webrtc-review.googlesource.com/74920
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23161}
2018-05-08 08:00:45 +00:00
Henrik Lundin
f8ed5614c2 Remove last mention of speex codec
The last mention was in a unit test, where speex was used to name an
arbitrary codec. The name "foo" is now used instead.

Bug: webrtc:4844
Change-Id: Ia1ede8512b894e6c16c0c168a50dc4d62d6911ad
Reviewed-on: https://webrtc-review.googlesource.com/74781
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23160}
2018-05-08 06:16:05 +00:00
Autoroller
092bbb5193 Roll chromium_revision 972ad081bc..5de2157f1e (556577:556680)
Change log: 972ad081bc..5de2157f1e
Full diff: 972ad081bc..5de2157f1e

Changed dependencies:
* src/base: 7e15f09288..c5356a4b51
* src/ios: 7cf6557612..9d27efb09d
* src/testing: fc9bddc533..f0ade05cb2
* src/third_party: 6cffa789bd..62736ae0ad
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e54b6ffb6d..e853531767
* src/third_party/depot_tools: 5e5f2d6035..5ae86d2021
* src/tools: 6fac6ce82d..abafe60c0f
DEPS diff: 972ad081bc..5de2157f1e/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I5635c961d38d01bbed1968607fb5246e88574c1a
Reviewed-on: https://webrtc-review.googlesource.com/75102
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23159}
2018-05-08 04:39:05 +00:00
Autoroller
8e952d1f1c Roll chromium_revision ba34164cc1..972ad081bc (556466:556577)
Change log: ba34164cc1..972ad081bc
Full diff: ba34164cc1..972ad081bc

Changed dependencies:
* src/base: 12029651a6..7e15f09288
* src/build: a163f9bb29..b61b6b6a2e
* src/ios: 94d3e02398..7cf6557612
* src/testing: b60c0be350..fc9bddc533
* src/third_party: 4b645d28cd..6cffa789bd
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b9ff578f32..e54b6ffb6d
* src/third_party/googletest/src: a6f06bf2fd..045e7f9ee4
* src/third_party/libvpx/source/libvpx: e4408a07be..28801f91c4
* src/tools: be60a7465b..6fac6ce82d
DEPS diff: ba34164cc1..972ad081bc/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I38f6342ddaac10ee8f0422433b291d24695bc6f2
Reviewed-on: https://webrtc-review.googlesource.com/75020
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23158}
2018-05-07 22:38:04 +00:00
Steve Anton
3172c035d5 Implement OnRemoveTrack and OnRemoveStream for Unified Plan
Also parameterizes the PeerConnection RTP unit tests to test
Unified Plan also.

Bug: webrtc:8587
Change-Id: I7661d9f2ec4b3bce0d2e2979035fa02225e3f118
Reviewed-on: https://webrtc-review.googlesource.com/73284
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23157}
2018-05-07 20:51:28 +00:00
Steve Anton
31e5bf59e0 Remove deprecated SocketFactory overrides.
Bug: webrtc:9198
Change-Id: I0ec3e4120e936fefa76e6cef968d7f615f568aa8
Reviewed-on: https://webrtc-review.googlesource.com/73964
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23156}
2018-05-07 20:03:27 +00:00
Seth Hampson
31dbc246d7 Adding PeerConnection.Observer.onTrack to the Java SDK.
Bug: webrtc:8869
Change-Id: I4c33f9ddf293af8c093a8726431a3574ff2b6e39
Reviewed-on: https://webrtc-review.googlesource.com/73966
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23155}
2018-05-07 18:22:48 +00:00
braveyao
1f184f0a15 [desktopCapture Mac] have latest iosurface before invalidating region
Always having the latest iosurface before invalidating a region.
Otherwise if CaptureFrame() happens in between, the capture result
may not be fully refreshed. Also we can't add lock since it will
impact performance.

Bug: webrtc:8652
Change-Id: Ib23105b16065018c691685083b76a771ce8771d3
Reviewed-on: https://webrtc-review.googlesource.com/74643
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23154}
2018-05-07 17:52:08 +00:00
Autoroller
4a22921985 Roll chromium_revision 63c80f2e33..ba34164cc1 (556358:556466)
Change log: 63c80f2e33..ba34164cc1
Full diff: 63c80f2e33..ba34164cc1

Changed dependencies:
* src/base: 6772872e6c..12029651a6
* src/build: 1fd2d08a6f..a163f9bb29
* src/ios: 2a4cc722a7..94d3e02398
* src/testing: 9c2c516d0f..b60c0be350
* src/third_party: b29bfab599..4b645d28cd
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3059fd7dba..b9ff578f32
* src/tools: 1293f814ef..be60a7465b
DEPS diff: 63c80f2e33..ba34164cc1/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Iad0237bd2f42d69348d93af0f0b2d4cbcfe7fbb3
Reviewed-on: https://webrtc-review.googlesource.com/74940
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23153}
2018-05-07 17:22:48 +00:00
Alessio Bazzica
d5ef6ff258 Disable WavWriterTest flaky tests on Mac.
TBR=titovartem@webrtc.org

Bug: webrtc:9247
Change-Id: I3d01ac5dd7d6ac2ff83f2b991238ce003c0182e9
Reviewed-on: https://webrtc-review.googlesource.com/74880
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23152}
2018-05-07 17:14:09 +00:00
Minyue Li
27e2b7d177 Plot NetEq stats in RTC event log visualizer.
Bug: webrtc:9147
Change-Id: I61ec7bc5299201e25e1efc503b73b84d5be3ebbf
Reviewed-on: https://webrtc-review.googlesource.com/71740
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23151}
2018-05-07 17:01:48 +00:00
Alessio Bazzica
0424c19fda AGC2 RNN VAD: FFT utility lib
BandAnalysisFft class that wraps the FFT library, makes it easy to change
FFT library, applies windowing function and owns the FFT input buffer.

Bug: webrtc:9076
Change-Id: I9e7ed587ae263b906e04a66bf8c06eaae64daf19
Reviewed-on: https://webrtc-review.googlesource.com/72900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23150}
2018-05-07 16:16:28 +00:00
Sebastian Jansson
bd9fe092ce Using shared task queue for congestion controller.
This simplifies the code and removes the need for a lot of bookkeeping
variables.

Bug: webrtc:9232
Change-Id: I0c9a4b0741ed5353caa22ba5acdcb166357441f2
Reviewed-on: https://webrtc-review.googlesource.com/74240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23149}
2018-05-07 15:54:38 +00:00