Commit Graph

11067 Commits

Author SHA1 Message Date
5e5f7e14b2 Remove unneeded enum forward declaration
While building Chrome with the VC++ 2017 /permissive- flag I got a
warning about a forward declaration of enum RateControlRegion. Untyped
forward declarations of enums are illegal because the compiler doesn't
know what size to make them. The only reason this forward declaration is
legal is because it isn't needed (the type is already defined).

This was found because /permissive- (or, equivalently for this purpose,
/w14471) incorrectly fires on this forward declaration even though it is
legal.

BUG=chromium:736059

Review-Url: https://codereview.webrtc.org/2834753002
Cr-Commit-Position: refs/heads/master@{#18741}
2017-06-24 20:04:29 +00:00
e5960ce737 Revert "Revert "Revert "Revert "Support more formats in RTCVideoFrame""""
This reverts commit 1cfeb435427a2fa677a495e34c882096efc193d0.

Reason for revert: Fix unit test

Original change's description:
> Revert "Revert "Revert "Support more formats in RTCVideoFrame"""
> 
> This reverts commit 7583390d1a3a7c4e9a77da0d77250abac0c34d1d.
> 
> Reason for revert: Breaks unit tests
> 
> Original change's description:
> > Revert "Revert "Support more formats in RTCVideoFrame""
> > 
> > This reverts commit 0789dab2cbd1617e94d7300e375163d42345f3d4.
> > 
> > Reason for revert: Include obc_corevideoframebuffer target
> > 
> > Original change's description:
> > > Revert "Support more formats in RTCVideoFrame"
> > > 
> > > This reverts commit bd2220a9c496ef2e8567b68d4be9435a110bdc34.
> > > 
> > > Reason for revert: Broke external clients
> > > 
> > > Original change's description:
> > > > Support more formats in RTCVideoFrame
> > > > 
> > > > Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
> > > > RTCVideoFrame.
> > > > 
> > > > Bug: webrtc:7785
> > > > Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
> > > > Reviewed-on: https://chromium-review.googlesource.com/536773
> > > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#18691}
> > > 
> > > TBR=magjed@webrtc.org,andersc@webrtc.org
> > > 
> > > Change-Id: Id765dd9543ed0613a6b2de108b268c3501025fcd
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: webrtc:7785
> > > Reviewed-on: https://chromium-review.googlesource.com/542837
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#18697}
> > 
> > TBR=magjed@webrtc.org,andersc@webrtc.org
> > 
> > Change-Id: I1ef5313b4a6c56eb8c7fd02d95db62c4e3c00255
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:7785
> > Reviewed-on: https://chromium-review.googlesource.com/542838
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#18716}
> 
> TBR=magjed@webrtc.org,andersc@webrtc.org
> 
> Change-Id: Id12f33698eb02041607cb9a5c54f37f01bfac5b1
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7785
> Reviewed-on: https://chromium-review.googlesource.com/544840
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18718}

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

Change-Id: I184303ecba8db91ef7de709f982a295a2efe92eb
Bug: webrtc:7785
Reviewed-on: https://chromium-review.googlesource.com/544841
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18731}
2017-06-23 10:59:41 +00:00
23cc468ddf Return WrappedI444Buffer in VP9Impl
This CL allows VP9Impl to return a WrappedI444Buffer if the image format is
VPX_IMG_FMT_I444.

Needs to be checked in after Chromium CL 2876363003 is checked in.

BUG=webrtc:7506

Review-Url: https://codereview.webrtc.org/2927943003
Cr-Commit-Position: refs/heads/master@{#18727}
2017-06-23 03:28:06 +00:00
e92f93fc9a Test picture sequence id when VideoSendStream is recreated.
Bug: webrtc:7475
Change-Id: I613b20b9da2c264b05c6a604d3f9754878857a0f
Reviewed-on: https://chromium-review.googlesource.com/527076
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18722}
2017-06-22 15:57:10 +00:00
4d25a0554a Disable RTCVideoRenderFrameCallbackNV12 test
It has been crashing on trybots

BUG=webrtc:7863
TBR=magjed@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2951273002
Cr-Commit-Position: refs/heads/master@{#18720}
2017-06-22 15:19:01 +00:00
1cfeb43542 Revert "Revert "Revert "Support more formats in RTCVideoFrame"""
This reverts commit 7583390d1a3a7c4e9a77da0d77250abac0c34d1d.

Reason for revert: Breaks unit tests

Original change's description:
> Revert "Revert "Support more formats in RTCVideoFrame""
> 
> This reverts commit 0789dab2cbd1617e94d7300e375163d42345f3d4.
> 
> Reason for revert: Include obc_corevideoframebuffer target
> 
> Original change's description:
> > Revert "Support more formats in RTCVideoFrame"
> > 
> > This reverts commit bd2220a9c496ef2e8567b68d4be9435a110bdc34.
> > 
> > Reason for revert: Broke external clients
> > 
> > Original change's description:
> > > Support more formats in RTCVideoFrame
> > > 
> > > Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
> > > RTCVideoFrame.
> > > 
> > > Bug: webrtc:7785
> > > Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
> > > Reviewed-on: https://chromium-review.googlesource.com/536773
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#18691}
> > 
> > TBR=magjed@webrtc.org,andersc@webrtc.org
> > 
> > Change-Id: Id765dd9543ed0613a6b2de108b268c3501025fcd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:7785
> > Reviewed-on: https://chromium-review.googlesource.com/542837
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#18697}
> 
> TBR=magjed@webrtc.org,andersc@webrtc.org
> 
> Change-Id: I1ef5313b4a6c56eb8c7fd02d95db62c4e3c00255
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7785
> Reviewed-on: https://chromium-review.googlesource.com/542838
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18716}

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

Change-Id: Id12f33698eb02041607cb9a5c54f37f01bfac5b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7785
Reviewed-on: https://chromium-review.googlesource.com/544840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18718}
2017-06-22 13:06:45 +00:00
d0fc37a884 Allow parsing empty RTCP TargetBitrate messages, but stop sending them.
Also, add ToString() convenience method to the target bitrate struct. Super useful when doing printf debugging :)

BUG=webrtc:7858

Review-Url: https://codereview.webrtc.org/2947633003
Cr-Commit-Position: refs/heads/master@{#18717}
2017-06-22 12:40:25 +00:00
7583390d1a Revert "Revert "Support more formats in RTCVideoFrame""
This reverts commit 0789dab2cbd1617e94d7300e375163d42345f3d4.

Reason for revert: Include obc_corevideoframebuffer target

Original change's description:
> Revert "Support more formats in RTCVideoFrame"
> 
> This reverts commit bd2220a9c496ef2e8567b68d4be9435a110bdc34.
> 
> Reason for revert: Broke external clients
> 
> Original change's description:
> > Support more formats in RTCVideoFrame
> > 
> > Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
> > RTCVideoFrame.
> > 
> > Bug: webrtc:7785
> > Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
> > Reviewed-on: https://chromium-review.googlesource.com/536773
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#18691}
> 
> TBR=magjed@webrtc.org,andersc@webrtc.org
> 
> Change-Id: Id765dd9543ed0613a6b2de108b268c3501025fcd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7785
> Reviewed-on: https://chromium-review.googlesource.com/542837
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18697}

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

Change-Id: I1ef5313b4a6c56eb8c7fd02d95db62c4e3c00255
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7785
Reviewed-on: https://chromium-review.googlesource.com/542838
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18716}
2017-06-22 12:23:03 +00:00
57ff3f4ec8 Remove aec_dump_unittests from audio_processing_tests.
It was included twice. In both of these targets:
webrtc/modules/audio_processing:{audio_processing_tests,
     audio_processing_unittests}

In audio_processing_tests, the new unit tests were added to 
public_deps, which (we think) somehow caused webrtc:webrtc_tests
to list the AecDump tests, to much confusion. 

Bug: webrtc:7404
Change-Id: I5788d93fef00d30a523312f317dde90eb64db8de
Reviewed-on: https://chromium-review.googlesource.com/543120
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18715}
2017-06-22 12:18:51 +00:00
28d5258d84 Force value-initialization of array to avoid uninitialized values.
An automatic undef checker noticed an error in the new
audio_processing_fuzzer. An array was default-initialized, meaning its
contents had indeterminate values. This array was later filtered for
forbidden values, which happened or did not happen based on the
uninitialized values. Hence the error.

TBR=henrik.lundin@webrtc.org

NOTRY=true

Bug: 735718, webrtc:7820
Change-Id: Ib50a6f7a5f054472fef07a27d37e02cea432a8ca
Reviewed-on: https://chromium-review.googlesource.com/544937
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18713}
2017-06-22 10:57:11 +00:00
300ec8c8db Remove WEBRTC_TRACE from webrtc/modules/audio_coding
We'd like to remove all occurrences of WEBRTC_TRACE and delete the
macro! One logging mechanism is enough.


NOTRY=True

Bug: webrtc:5118
Change-Id: Ic226318e0aebe3a71785fcb4ce07371872ab7128
Reviewed-on: https://chromium-review.googlesource.com/518133
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18712}
2017-06-22 10:05:51 +00:00
1d29c86cbf Make VideoProcessor::Init void (always returning true).
BUG=none

Review-Url: https://codereview.webrtc.org/2946263002
Cr-Commit-Position: refs/heads/master@{#18711}
2017-06-22 09:18:50 +00:00
451975206a Enable more unittests on iOS, and disable those that fail on simulator
Tests enabled:
* modules_unittests
* ortc_unittests
* rtc_media_unittests
* rtc_unittests
* video_capture_tests
* video_engine_tests

BUG=webrtc:5566,webrtc:4752,webrtc:5568,webrtc:5569

Review-Url: https://codereview.webrtc.org/2938193002
Cr-Commit-Position: refs/heads/master@{#18710}
2017-06-22 08:47:20 +00:00
ab97e18fa9 Fix the binary size regression on Chromium Windows.
There is a dependency chain from Chromium windows main_dll to Opus
which should never exist. We used to rely on rtc_static_library
to break this chain. So this CL replaced some rtc_source_set
with rtc_static_library.

libvpx fix (https://chromium-review.googlesource.com/c/544107/) for
ios-simulator linking issue is landed and this CL can be sumbitted once the new
Chromium is rolled into WebRTC.

BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2947273002
Cr-Commit-Position: refs/heads/master@{#18709}
2017-06-22 08:28:59 +00:00
c6de0c98af Upgrade to (D)TLS1.2 using the new BoringSSL (D)TLSv1_2_method functions
Bug: webrtc:7865
Change-Id: I39344f385181132fe2e0f832eec1cf8fe0736dfe
Reviewed-on: https://chromium-review.googlesource.com/543795
Commit-Queue: Emad Omara <emadomara@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18706}
2017-06-22 00:30:04 +00:00
1168fd4ed5 What can't loopback test be more like full stack test?
It can; this CL makes it a lot closer, if not all the way to a merge.
Performance from video_loopback and screenshare_loopback should now
match what we're seeing in FullStackTest, which will make debugging and
assesment of quality differences much easier.

It also adds the ability to view all of the simulcast streams at once,
in separate windows.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2946893003
Cr-Commit-Position: refs/heads/master@{#18703}
2017-06-21 16:00:17 +00:00
10894996ef Fix timing frames and FEC conflict
Reenable pacer_exit timestamp updates for the timing frames and
exclude timing-frames carrying packets from the FEC.

BUG=webrtc:7859

Review-Url: https://codereview.webrtc.org/2947133002
Cr-Commit-Position: refs/heads/master@{#18702}
2017-06-21 15:23:19 +00:00
83c97da593 Only append SPS/PPS to bitstream if supplied out of band.
BUG=chromium:721597

Review-Url: https://codereview.webrtc.org/2945853002
Cr-Commit-Position: refs/heads/master@{#18701}
2017-06-21 14:22:40 +00:00
73e2180f94 Add webrtc/rtc_base skeleton.
Add webrtc/rtc_base in preparation for the larger rename
in https://codereview.webrtc.org/2877023002.
This is needed in order to make downstream projects update
their dependencies (since it's not possible to use public_deps
for android_library).

BUG=webrtc:7634
NOTRY=True

Change-Id: I90fa72c9ebbc4e69bb3133413c3258432a5a72b1
Reviewed-on: https://chromium-review.googlesource.com/541215
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18699}
2017-06-21 12:50:22 +00:00
1b2469b878 Fix AVFoundation framework import
When building the WebRTC project for iOS, the build will fail on Xcode 9
because of a missing framework-header (AVFoundation). This pull-request
will add the missing "#import <AVFoundation/AVFoundation.h>" line to the
"RTCCameraVideoCapturer" class.

BUG=webrtc:7846

Review-Url: https://codereview.webrtc.org/2944753002
Cr-Commit-Position: refs/heads/master@{#18698}
2017-06-21 10:44:05 +00:00
0789dab2cb Revert "Support more formats in RTCVideoFrame"
This reverts commit bd2220a9c496ef2e8567b68d4be9435a110bdc34.

Reason for revert: Broke external clients

Original change's description:
> Support more formats in RTCVideoFrame
> 
> Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
> RTCVideoFrame.
> 
> Bug: webrtc:7785
> Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
> Reviewed-on: https://chromium-review.googlesource.com/536773
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18691}

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

Change-Id: Id765dd9543ed0613a6b2de108b268c3501025fcd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7785
Reviewed-on: https://chromium-review.googlesource.com/542837
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18697}
2017-06-21 08:41:37 +00:00
0f15f926e3 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
And implementation class RtpStreamReceiverController.
It's responsible for demuxing, and acts as factory for
RtpStreamReceiverInterface.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2886993005
Cr-Commit-Position: refs/heads/master@{#18696}
2017-06-21 08:05:22 +00:00
130ca7e783 Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )
Reason for revert:
Relanding the orginal CL. The breakage would be a flakey build.

Original issue's description:
> Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ )
>
> Reason for revert:
> The Android 32 (more config) bot is broken.
>
> Original issue's description:
> > Try to fix the binary size increase issue on Chromium.
> >
> > The target common_video used to depend on rtc_media_base which introduces
> > the dependency on p2p. This probably causes the binary size increase on Win
> > Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
> >
> > BUG=chromium:734631
> >
> > Review-Url: https://codereview.webrtc.org/2945233002
> > Cr-Commit-Position: refs/heads/master@{#18693}
> > Committed: 9ed1609737
>
> TBR=kjellander@webrtc.org,deadbeef@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2949953003
> Cr-Commit-Position: refs/heads/master@{#18694}
> Committed: c2e208a924

TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2949883003
Cr-Commit-Position: refs/heads/master@{#18695}
2017-06-21 08:02:59 +00:00
c2e208a924 Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ )
Reason for revert:
The Android 32 (more config) bot is broken.

Original issue's description:
> Try to fix the binary size increase issue on Chromium.
>
> The target common_video used to depend on rtc_media_base which introduces
> the dependency on p2p. This probably causes the binary size increase on Win
> Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.
>
> BUG=chromium:734631
>
> Review-Url: https://codereview.webrtc.org/2945233002
> Cr-Commit-Position: refs/heads/master@{#18693}
> Committed: 9ed1609737

TBR=kjellander@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2949953003
Cr-Commit-Position: refs/heads/master@{#18694}
2017-06-21 07:30:49 +00:00
9ed1609737 Try to fix the binary size increase issue on Chromium.
The target common_video used to depend on rtc_media_base which introduces
the dependency on p2p. This probably causes the binary size increase on Win
Chromium. Some chromium targets like src/media/gpu:gpu depends on common_video directly.

BUG=chromium:734631

Review-Url: https://codereview.webrtc.org/2945233002
Cr-Commit-Position: refs/heads/master@{#18693}
2017-06-21 06:58:36 +00:00
37aa8ba616 Test and fix for huge bwe drop after alr state.
BUG=webrtc:7746

Review-Url: https://codereview.webrtc.org/2931873002
Cr-Commit-Position: refs/heads/master@{#18692}
2017-06-21 06:42:30 +00:00
bd2220a9c4 Support more formats in RTCVideoFrame
Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
RTCVideoFrame.

Bug: webrtc:7785
Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
Reviewed-on: https://chromium-review.googlesource.com/536773
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18691}
2017-06-21 06:30:23 +00:00
38145241a2 Create VideoDecoderFactory interface and implementation.
The implementation creates an Android hardware video decoder.  It is built
around the same patterns as the HardwareVideoEncoderFactory.

This change pulls some shared code and constants into a common "utils" class.

Finally, adds an instrumentation test for the HardwareVideoDecoder.

BUG=webrtc:7760

Change-Id: Iea6eaae7727925743cb54f7c3153a6c07d62f55d
Reviewed-on: https://chromium-review.googlesource.com/536254
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18686}
2017-06-20 18:37:30 +00:00
b080b46df4 Create a hardware VideoDecoder implementation using Android MediaCodec.
BUG=webrtc:7760

Change-Id: Ieae3852d22cadf24cf4184ae985062918a85f02c
Reviewed-on: https://chromium-review.googlesource.com/536237
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18685}
2017-06-20 18:04:08 +00:00
fde2116288 Use constexpr to avoid a static initializer
Floating-point calculations are not guaranteed to happen at compile time
unless you force the issue with constexpr. This initializer was found
by running tools\win\static_initializers on a canary build
chrome_child.dll. constexpr was added to kSilenceRms for consistency.

BUG=chromium:341941

Review-Url: https://codereview.webrtc.org/2943833002
Cr-Commit-Position: refs/heads/master@{#18684}
2017-06-20 17:57:09 +00:00
a2af000882 Improve the simulation stats aggregation in neteq_rtpplay
The network stats used to be polled from the NetEq object once at the
very end of the simulation. With this change, the stats are polled
once every second, and then aggregated at the end of the run. This
leads to more meaningful numbers.

Bug: webrtc:2692
Change-Id: I9e0f4ddada2f9e42fb9234970deb1af235fffc8c
Reviewed-on: https://chromium-review.googlesource.com/541218
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18682}
2017-06-20 16:20:00 +00:00
2b3e061443 Hotfix for psnr regresion with fec tests caused by timing frames.
BUG=chromium:735001,webrtc:7594

Review-Url: https://codereview.webrtc.org/2946893002
Cr-Commit-Position: refs/heads/master@{#18681}
2017-06-20 15:52:27 +00:00
0bc0ccdc43 Add Matlab plotting script generator to neteq_rtpplay
This change adds an option to have neteq_rtpplay generate a Matlab
script. When executed in Matlab, the script will generate graphs with
the timing information from the test run.

The script is generated when the flag --matlabplot is passed to
neteq_rtpplay.

The CL also adds better checking and reporting about packets discarded
in the process of finding out the initial sampling rate.

Bug: webrtc:2692, webrtc:7467
Change-Id: I805e7c83b82533142b6e74bf065506e3d60a8170
Reviewed-on: https://chromium-review.googlesource.com/541276
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18680}
2017-06-20 14:22:19 +00:00
8c51282f7f Added new AudioProcessing fuzzer
This is a high-level fuzzer that creates an AudioProcessing instance.
All possible combinations of publicly visible components are
fuzzed. Input and output sample rate, call order and use of the
float/fix interface is fuzzed. Sample rate may change between calls.

To fuzz floating point numbers, raw data is converted to floats, and
filtered for special values like 'inf', 'nan' and very large
values. Note that the default use case of APM is to only allow values
between +/- 2^15.

BUG=webrtc:7820

Review-Url: https://codereview.webrtc.org/2876793002
Cr-Commit-Position: refs/heads/master@{#18678}
2017-06-20 12:26:55 +00:00
be45757028 Add henrik.lundin to fuzzers OWNERS
NOTRY=True

Bug: None
Change-Id: Icc7137b2072ac6bbdf45edf815b9260bb4e7169d
Reviewed-on: https://chromium-review.googlesource.com/541279
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18677}
2017-06-20 11:58:43 +00:00
c8ece43a62 Minor updates to VideoReceiveStream:
* Change decoder thread to use new thread function type.
* Reduce the time of when video_receiver_ receives callbacks on the process thread to match with Start/Stop of the decoder.
* Not triggering shutdown unless the thread is running.

BUG=webrtc:7361

Review-Url: https://codereview.webrtc.org/2944033003
Cr-Commit-Position: refs/heads/master@{#18675}
2017-06-20 09:44:38 +00:00
a9b848abc9 Bugfix:setting capture framerate always defaults to 30fps.
Bug: webrtc:7777
Change-Id: Ia8db06eea29d271fe8a74db203b68cabd4fa5939
Reviewed-on: https://chromium-review.googlesource.com/539641
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18673}
2017-06-20 07:50:39 +00:00
cadd306857 Fix test break by the recent changes in IcerServer
Bug: webrtc:6973
Change-Id: I43ab49831fc3fe653eb26739a1ef3a60c73b9ffd
Reviewed-on: https://chromium-review.googlesource.com/540286
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Commit-Queue: Emad Omara <emadomara@google.com>
Cr-Commit-Position: refs/heads/master@{#18670}
2017-06-19 22:48:53 +00:00
6dd77c4d89 Add reference counter of DxgiDuplicatorController to unload DXGI components
On Windows, only four applications can use DXGI duplication APIs concurrently.
So this change adds a reference counter of DxgiDuplicatorController to unload
DXGI components when the reference counter reaches 0.

BUG=webrtc:7808

Review-Url: https://codereview.webrtc.org/2933893003
Cr-Commit-Position: refs/heads/master@{#18668}
2017-06-19 20:59:42 +00:00
3352ce92f9 Android: Modular WebRTC follow-up
This CL cleans up parts from https://codereview.webrtc.org/2939203002/.

Bug: webrtc:7613
Change-Id: I96d1a2cc91174f43d3cae2cb41b2e0fe7142e3e9
Reviewed-on: https://chromium-review.googlesource.com/539456
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18665}
2017-06-19 16:05:53 +00:00
42308f615c Fix uploading of available send bitrate statistics.
BUG=webrtc:5079
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2943073002 .
Cr-Commit-Position: refs/heads/master@{#18664}
2017-06-19 15:58:15 +00:00
ce433fafc1 Revert "Adding ANA config event to debug dump."
This reverts commit 652abc9a472426367e149db5a101b894179687aa.

Reason for revert: break upstream bots

Original change's description:
> Adding ANA config event to debug dump.
> 
> BUG=webrtc:7854
> 
> Change-Id: I12c33b8558fd49374a55282c391b87fde9e13a28
> Reviewed-on: https://chromium-review.googlesource.com/535554
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Michael T <tschumim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#18661}

TBR=minyue@webrtc.org,ossu@webrtc.org,tschumim@webrtc.org

Change-Id: Id1f93338e431c9cd8dade722be7edd16a648d044
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7854
Reviewed-on: https://chromium-review.googlesource.com/539737
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18663}
2017-06-19 15:23:02 +00:00
bfe45c29c5 Use uint8 pointer instead of std::vector in NV12Scale.
To prepare for landing 536773.

Bug: webrtc:7785
Change-Id: I841218dca3fb9d83f362f7f2b9076f3f189e7c15
Reviewed-on: https://chromium-review.googlesource.com/539577
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18662}
2017-06-19 15:08:48 +00:00
652abc9a47 Adding ANA config event to debug dump.
BUG=webrtc:7854

Change-Id: I12c33b8558fd49374a55282c391b87fde9e13a28
Reviewed-on: https://chromium-review.googlesource.com/535554
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18661}
2017-06-19 15:00:39 +00:00
3093ef193e Android JNI: Clean up AndroidVideoTrackSource and NativeHandleImpl
I'm preparing adding support for Java VideoFrames in
AndroidVideoTrackSource. I split out small unrelated clean-ups into this
CL in order to make the big CL more focused.

Bug: webrtc:7749
Change-Id: Ib261ab8eb055898b39307d4e78935bf60d323820
Reviewed-on: https://chromium-review.googlesource.com/539638
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18660}
2017-06-19 14:46:08 +00:00
04f4d126f8 Implement timing frames.
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
2017-06-19 14:18:55 +00:00
1f7476f7a3 Remove explicit draw call on MTKView.
Bug: webrtc:7852
Change-Id: I21dc1446d6c456c8750a1d80bd1facddfff50ab8
Reviewed-on: https://chromium-review.googlesource.com/539336
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18657}
2017-06-19 13:08:04 +00:00
91047e566e Remove redundant std::min from ProbeBitrateEstimator.
Mimimum was already computed on line 139.

BUG=None

Review-Url: https://codereview.webrtc.org/2945833002
Cr-Commit-Position: refs/heads/master@{#18656}
2017-06-19 13:07:30 +00:00
bed7a6b3b6 Use information about blacklisted devices in video_quality_loopback_test
BUG=webrtc:7724
NOTRY=True

Review-Url: https://codereview.webrtc.org/2937123002
Cr-Commit-Position: refs/heads/master@{#18654}
2017-06-19 08:16:45 +00:00
1a610f15c3 Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )
Reason for revert:
Breaking google3 projects

Original issue's description:
> Opus implementation of the AudioEncoderFactoryTemplate API
>
> Now the templated AudioEncoderFactory can create Opus encoders!
>
> BUG=webrtc:7831
>
> Review-Url: https://codereview.webrtc.org/2930243003
> Cr-Commit-Position: refs/heads/master@{#18645}
> Committed: fe1aa82c63

TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7831

Review-Url: https://codereview.webrtc.org/2947563002
Cr-Commit-Position: refs/heads/master@{#18649}
2017-06-18 09:38:58 +00:00