11375 Commits

Author SHA1 Message Date
danilchap
901b2df431 Simplify FakeReceiveStatistics in video send stream tests
Rtcp sender now take smaller interface making it possible to simplify the fake

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2984283002
Cr-Commit-Position: refs/heads/master@{#19181}
2017-07-28 15:56:04 +00:00
ehmaldonado
35a872c0e6 Make RTCStatsReport::ToString() return JSON-parseable string.
BUG=chromium:653087

Review-Url: https://codereview.webrtc.org/2983243002
Cr-Commit-Position: refs/heads/master@{#19180}
2017-07-28 14:29:12 +00:00
sakal
836f60cda1 Move matrix from VideoFrame to TextureBuffer.
Previously, the matrix in VideoFrame was used to crop and scale the
frame. This caused complications because webrtc::VideoFrame doesn't
include a matrix. cropAndScale method is added to VideoBuffer class for
cropping and scaling instead.

BUG=webrtc:7749, webrtc:7760

Review-Url: https://codereview.webrtc.org/2990583002
Cr-Commit-Position: refs/heads/master@{#19179}
2017-07-28 14:12:23 +00:00
tschumim
54348fb5ce Removed an obsolete DCHECK in AudioEncoderOpus.
BUG=None

Review-Url: https://codereview.webrtc.org/2986083002
Cr-Commit-Position: refs/heads/master@{#19177}
2017-07-28 09:52:59 +00:00
eladalon
eaec118240 Remove DCHECK from Call's ctor that could never fail
I don't think this line could never conceivably fail - if the ctor has reached that point, the object fit in memory, and its members have all been allocated legal memory addresses, none of which may be 0x00.

BUG=None

Review-Url: https://codereview.webrtc.org/2989813002
Cr-Commit-Position: refs/heads/master@{#19176}
2017-07-28 09:25:09 +00:00
deadbeef
4cd599f025 If adapter type is unknown and interface name is "ipsec", treat as VPN.
This will result in the ipsec interfaces being prioritized below Wi-Fi
and cell interfaces. This makes the most difference when we hit the
default limit for IPv6 interfaces (5), and there are lots of ipsec
interfaces for whatever reason, resulting in the "real" interfaces that
would actually succeed not being used. See the linked bug 7703.

BUG=webrtc:7703, webrtc:3149

Review-Url: https://codereview.webrtc.org/2985133002
Cr-Commit-Position: refs/heads/master@{#19175}
2017-07-27 22:05:29 +00:00
jbauch
4c27a96767 Remove libsrtp 2.0.0 compatibility code.
The upgrade to libsrtp 2.1.0 rolled in https://codereview.webrtc.org/2968463002
so the compatibility code can be removed.

BUG=webrtc:7856

Review-Url: https://codereview.webrtc.org/2969543002
Cr-Commit-Position: refs/heads/master@{#19174}
2017-07-27 22:04:20 +00:00
minyue-webrtc
516711cde9 Turning on Opus 120ms frame length switch.
Chromium has adopted Opus 1.2.1 which allows 120ms frame encoding. It
is time to turn on the switch for building WebRTC with this feature.


Bug: webrtc:8042
TBR: kjellander@webrtc.org
Change-Id: I644b47cfb56f835695ef1263741cda6e3ee3d862
Reviewed-on: https://chromium-review.googlesource.com/586725
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Felicia Lim <flim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19173}
2017-07-27 17:23:35 +00:00
deadbeef
28e2919cfd Adding Android binding for RTCConfiguration::max_ipv6_networks.
BUG=webrtc:7703

Review-Url: https://codereview.webrtc.org/2984863002
Cr-Commit-Position: refs/heads/master@{#19172}
2017-07-27 16:14:38 +00:00
sakal
9eb3d19ec0 Fix a crash in PeerConnectionFactory.SetVideoHwAccelerationOptions.
BUG=webrtc:8035

Review-Url: https://codereview.webrtc.org/2992523002
Cr-Commit-Position: refs/heads/master@{#19171}
2017-07-27 15:23:58 +00:00
sakal
2d4040ed0e Add a comment that RTCAVFoundationVideoSource is deprecated.
RTCAVFoundationVideoSource is deprecated and will removed after a few
weeks.

BUG=webrtc:7177
R=magjed@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2992613002
Cr-Commit-Position: refs/heads/master@{#19170}
2017-07-27 14:48:57 +00:00
minyue
81f1da3dd0 Adding missing resources to audio_codec_speed_tests.
BUG=none

Review-Url: https://codereview.webrtc.org/2727973004
Cr-Commit-Position: refs/heads/master@{#19168}
2017-07-27 12:49:57 +00:00
danilchap
f5f793c2ed Take smaller interface for RtpRtcp::Configuration::receive_statistics
BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2988763002
Cr-Commit-Position: refs/heads/master@{#19167}
2017-07-27 11:44:18 +00:00
philipel
77415f561d Revert of Disable SeqNumUnwrapper death tests to avoid breaking downstream builds. (patchset #1 id:1 of https://codereview.chromium.org/2985083002/ )
Reason for revert:
Creating revert to fix these tests.

Original issue's description:
> Disable SeqNumUnwrapper death tests to avoid breaking downstream builds.
>
> BUG=None
> TBR=stefan@webrtc.org
> NOTRY=true
>
> Review-Url: https://codereview.webrtc.org/2985083002
> Cr-Commit-Position: refs/heads/master@{#19155}
> Committed: 8e245561f2

TBR=stefan@webrtc.org
BUG=None

Review-Url: https://codereview.webrtc.org/2992643002
Cr-Commit-Position: refs/heads/master@{#19166}
2017-07-27 11:37:18 +00:00
minyue-webrtc
adb58b88a1 Renable some Opus tests after Opus 1.2.1 update.
Bug: webrtc:8024
Change-Id: Ia7b9de70ef85e4ac32a7b84088b79cc6a260cc69
Reviewed-on: https://chromium-review.googlesource.com/586867
Reviewed-by: Felicia Lim <flim@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19164}
2017-07-27 07:40:14 +00:00
Zijie He
9c0914f938 Do not crop DesktopFrame if the size won't change
CreateCroppedDesktopFrame() does not need to create a CroppedDesktopFrame if the
size won't change.

Bug: webrtc:8039
Change-Id: Ie6789a4b473b69bced94c4a25a68f1da6bb3510e
Reviewed-on: https://chromium-review.googlesource.com/587808
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19163}
2017-07-27 01:45:19 +00:00
deadbeef
2059bb3e4b Adding Obj-C binding for RTCConfiguration::max_ipv6_networks.
BUG=webrtc:7703

Review-Url: https://codereview.webrtc.org/2988553004
Cr-Commit-Position: refs/heads/master@{#19162}
2017-07-27 01:25:43 +00:00
Zijie He
ecf3d53088 Add histogram for FallbackDesktopCapturerWrapper and BlankDetectorDesktopCapturerWrapper
We should record the number of fallbacks and blank frames.

Bug: webrtc:8040
Change-Id: I92e7b7d7b4664fee6d6bd636609e80e532aa4bd4
Reviewed-on: https://chromium-review.googlesource.com/587688
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19161}
2017-07-27 00:55:59 +00:00
deadbeef
d21eab3eea Add "max_ipv6_networks" field to RTCConfiguration.
This allows an application to easily override the default limit
(currently 5).

Also adding a test that covers more of the
PeerConnection<->PortAllocator interaction.

BUG=webrtc:7703

Review-Url: https://codereview.webrtc.org/2985653003
Cr-Commit-Position: refs/heads/master@{#19160}
2017-07-26 23:50:11 +00:00
deadbeef
3427f538de Relanding: Move "max IPv6 networks" logic to BasicPortAllocator, and fix sorting.
Relanding because the broken chromium test has been fixed:
https://chromium-review.googlesource.com/582196

This CL moves the responsibility for restricting the number of IPv6
interfaces used for ICE to BasicPortAllocator. This is the right place
to do it in the first place; it's where all the rest of the filtering
occurs. And NetworkManager shouldn't need to know about ICE limitations;
only the ICE classes should.

Part of the reason I'm doing this is that I want to add a
"max_ipv6_networks" API to RTCConfiguration, so that applications can
override the default easily (see linked bug). But that means that
PeerConnection would need to be able to call "set_max_ipv6_networks" on
the underlying object that does the filtering, and that method isn't
available on the "NetworkManager" base class. So rather than adding
another method to a place it doesn't belong, I'm moving it to the place
it does belong.

In the process, I noticed that "CompareNetworks" is inconsistent with
"SortNetworks"; the former orders interfaces alphabetically, and the
latter reverse-alphabetically. I believe this was unintentional, and
results in undesirable behavior (like "eth1" being preferred over
"eth0"), so I'm fixing it and adding a test.

BUG=webrtc:7703

Review-Url: https://codereview.webrtc.org/2983213002
Cr-Original-Commit-Position: refs/heads/master@{#19112}
Committed: ad9561404c
Review-Url: https://codereview.webrtc.org/2983213002
Cr-Commit-Position: refs/heads/master@{#19159}
2017-07-26 23:09:33 +00:00
jbudorick
58f1725ff1 Add gn dependency between ana_debug_dump_proto and ana_config_proto.
BUG=chromium:746106

Review-Url: https://codereview.webrtc.org/2985853002
Cr-Commit-Position: refs/heads/master@{#19158}
2017-07-26 21:49:20 +00:00
Zijie He
74544f9d1b Return translated position in MouseCursorMonitor
This change returns translated position in the newly added overload
MouseCursorMonitor::Callback::OnMouseCursorPosition(DesktopVector) callback.

Meanwhile it also reduces the duplicate logic in Windows capturer
implementations. So except for the deprecated logic in MouseCursorMonitorWin,
all GetSystemMetrics() function calls are merged into GetScreenRect(),
GetFullscreenRect() and GetFullscreenTopLeft() functions.

Bug: webrtc:7950
Change-Id: Ic2a85a80b6947367bdd20d8f96f11e0f5c269006
Reviewed-on: https://chromium-review.googlesource.com/581951
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19157}
2017-07-26 20:38:14 +00:00
deadbeef
54c721541d Fix issues with NetworkMonitor singleton when used by multiple clients.
When you create multiple "PeerConnectionFactory"s, they end up using
the same NetworkMonitor singleton. But the second one's
"AndroidNetworkMonitor" class (in C++) wasn't getting the expected
network list update, and as a result it wasn't binding sockets to
networks successfully, acting as if the networks didn't exist.

The solution is just to move "updateActiveNetworkList" to
"startMonitoring". This CL also does some other minor
cleanup/refactoring, and fixes a more corner-casey issue where, if the
first PeerConnection is destroyed, the second one would stop receiving
network updates.

BUG=webrtc:7946

Review-Url: https://codereview.webrtc.org/2990693002
Cr-Commit-Position: refs/heads/master@{#19156}
2017-07-26 18:56:49 +00:00
philipel
8e245561f2 Disable SeqNumUnwrapper death tests to avoid breaking downstream builds.
BUG=None
TBR=stefan@webrtc.org
NOTRY=true

Review-Url: https://codereview.webrtc.org/2985083002
Cr-Commit-Position: refs/heads/master@{#19155}
2017-07-26 15:43:53 +00:00
philipel
7956c0f2f6 Implemented a new sequence number unwrapper in sequence_number_util.h.
There is already an Unwrapper in webrtc/modules/include/module_common_types.h,
but we reimplemented it in sequence_number_util.h for a few reasons:
 - Such a class belongs in sequence_number_util.h.
 - It is a cleaner implementation since we can use the rest of
   sequence_number_util.h functionality.
 - You can choose at which number the unwrapped sequence should start,
   which is used to avoid the edge case when a backward wrap can happen
   as the first few numbers are unwrapped.
 - This unwrapper can unwrap numbers that does not wrap 8/16/32 bits.

BUG=None

Review-Url: https://codereview.webrtc.org/2977603002
Cr-Commit-Position: refs/heads/master@{#19154}
2017-07-26 14:48:15 +00:00
minyue-webrtc
8de1826b6d Reland "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
BUG=b/63898232, b/64053465

Originally Reviewed-on: https://chromium-review.googlesource.com/584707

Reverted-on: https://chromium-review.googlesource.com/586268
Change-Id: I212b0c1e81a6ccd73b051e6728e601a8641463b8
Reviewed-on: https://chromium-review.googlesource.com/586328
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19153}
2017-07-26 14:28:51 +00:00
Minyue Li
7df370b69c Revert "Allow AudioSendStream to reconfig AudioNetworkAdaptor"
This reverts commit 4a88120e9568e48ba6e9b12045d56d745da2f34a.

Reason for revert: Found a mistake.

Original change's description:
> Allow AudioSendStream to reconfig AudioNetworkAdaptor
> 
> Bug: b/63898232, b/64053465
> Change-Id: I3485c35c0b74c0e2d654f8d70de0238a617a0ddc
> Reviewed-on: https://chromium-review.googlesource.com/584707
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Michael T <tschumim@webrtc.org>
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19150}

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

Change-Id: I7f6fdefac91bb119f528f117cb6ab6569202ee9a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/63898232, b/64053465
Reviewed-on: https://chromium-review.googlesource.com/586268
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19151}
2017-07-26 10:01:50 +00:00
minyue-webrtc
4a88120e95 Allow AudioSendStream to reconfig AudioNetworkAdaptor
Bug: b/63898232, b/64053465
Change-Id: I3485c35c0b74c0e2d654f8d70de0238a617a0ddc
Reviewed-on: https://chromium-review.googlesource.com/584707
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19150}
2017-07-26 09:48:59 +00:00
eladalon
abbc430ea0 Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.
BUG=None

Review-Url: https://codereview.webrtc.org/2987763003
Cr-Commit-Position: refs/heads/master@{#19149}
2017-07-26 09:09:44 +00:00
gyzhou
b38f38662f Update native plugin dll for turn servers and video.
This CL was modified from work of sharifferdous@ (intern supervised by lliuu@)

BUG=webrtc:7389

Review-Url: https://codereview.webrtc.org/2987723002
Cr-Commit-Position: refs/heads/master@{#19146}
2017-07-25 23:04:31 +00:00
jtteh
3b673c66a4 Removed file RTCCameraVideoCapturer.mm that isn't needed
Also added post commit review changes.

BUG=webrtc:7898

Review-Url: https://codereview.webrtc.org/2988783002
Cr-Commit-Position: refs/heads/master@{#19145}
2017-07-25 22:48:39 +00:00
peah
b1c9d1de36 Avoid that previous settings in APM are overwritten by WebRtcVoiceEngine
This CL ensures that any previously set nondefault settings in the
audio processing module are not overwritten by the ApplyOptions
method in WebRtcVoiceEngine

BUG=webrtc:8018

Review-Url: https://codereview.webrtc.org/2985633002
Cr-Commit-Position: refs/heads/master@{#19144}
2017-07-25 22:45:24 +00:00
jamiewalch
d1d6c5a31b Add jamiewalch to OWNERS.
BUG=None

Review-Url: https://codereview.webrtc.org/2989653002
Cr-Commit-Position: refs/heads/master@{#19142}
2017-07-25 21:37:07 +00:00
jtteh
61b0ed039d [iOS] Fix incorrectly oriented frames when rapidly switching between cameras.
During a call, with both phones in horizontal or landscape mode, rapidly switching between the front and back camera sometimes causes the remote video to be shown upside down.

There seems to be a race condition when setting the rotation based on the orientation of the device and which camera we're using.

So use the active input's camera to check instead of the client state.

BUG=webrtc:7898

Review-Url: https://codereview.webrtc.org/2964703002
Cr-Commit-Position: refs/heads/master@{#19139}
2017-07-25 16:35:55 +00:00
danilchap
96b69bdbee Refactor composing report blocks for rtcp Sender/Receiver reports.
Compose them while creating sr/rr instead of presaving in temporary
member variable

BUG=webrtc:5565, webrtc:8016

Review-Url: https://codereview.webrtc.org/2979413002
Cr-Commit-Position: refs/heads/master@{#19138}
2017-07-25 16:15:14 +00:00
eladalon
7fb11d7376 Shrink critical-section scope in ReceiveStatisticsImpl::GetActiveStatisticians()
The critical-section's scope can be shrunk (we can hold the lock for a shorter time).

BUG=None

Review-Url: https://codereview.webrtc.org/2984973002
Cr-Commit-Position: refs/heads/master@{#19137}
2017-07-25 15:25:23 +00:00
danilchap
6209dcdeb1 Add SetReportBlocks to rtcp Sender/Receive Report classes.
BUG=None

Review-Url: https://codereview.webrtc.org/2991623002
Cr-Commit-Position: refs/heads/master@{#19136}
2017-07-25 15:07:13 +00:00
kthelgason
fb143127d7 Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2980173002/ )
Reason for revert:
Relanding after fixing issues with no video.

Original issue's description:
> Revert of Injectable Obj-C video codecs (patchset #2 id:370001 of https://codereview.webrtc.org/2979983002/ )
>
> Reason for revert:
> Still having problems with no video. Reverting.
> Once no video is visible, no video is available from then on even if the callee app is in the foreground.
>
>
> Original issue's description:
> > Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2979973002/ )
> >
> > Reason for revert:
> > Fix the broken build file
> >
> > Original issue's description:
> > > Revert of Injectable Obj-C video codecs (patchset #3 id:400001 of https://codereview.webrtc.org/2981583002/ )
> > >
> > > Reason for revert:
> > > Breaks bots. Build file incorrect.
> > >
> > > Original issue's description:
> > > > Reland of Injectable Obj-C video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2975963002/ )
> > > >
> > > > Reason for revert:
> > > > New CL for fixing the issues
> > > >
> > > > Original issue's description:
> > > > > Revert of Injectable Obj-C video codecs (patchset #8 id:140001 of https://codereview.webrtc.org/2966023002/ )
> > > > >
> > > > > Reason for revert:
> > > > > Causes no video in certain scenarios. Please come up with a test plan or unit test to prevent such problems in the future.
> > > > >
> > > > > Original issue's description:
> > > > > > Injectable Obj-C video codecs
> > > > > >
> > > > > > Initial CL for this effort, with a working RTCVideoEncoder/Decoder for H264
> > > > > > (wrapping the VideoToolbox codec).
> > > > > >
> > > > > > Some notes / things left to do:
> > > > > >   - There are some hard-coded references to codec types that are supported by
> > > > > >     webrtc::VideoCodec, cricket::VideoCodec, webrtc::CodecSpecificInfo etc
> > > > > >     since we need to convert to/from these types in ObjCVideoEncoder/Decoder.
> > > > > >     These types would need to be more codec agnostic to avoid this.
> > > > > >   - Most interfaces are borrowed from the design document for injectable
> > > > > >     codecs in Android. Some data in the corresponding C++ classes is discarded
> > > > > >     when converting to the Obj-C version, since it has fewer fields. I have not
> > > > > >     verified whether all data that we do keep is needed, or whether we might be
> > > > > >     losing anything useful in these conversions.
> > > > > >   - Implement the VideoToolbox codec code directly in the RTCVideoEncoderH264
> > > > > >     classes, instead of wrapping webrtc::H264VideoToolboxEncoder / decoder.
> > > > > >     Eliminates converting between ObjC/C++ types outside the ObjCVideoEncoder/
> > > > > >     Decoder wrapper classes.
> > > > > >   - List the injected codec factory's supported codecs in the list of codecs in
> > > > > >     AppRTCMobile.
> > > > > >
> > > > > > BUG=webrtc:7924
> > > > > > R=magjed@webrtc.org
> > > > > >
> > > > > > Review-Url: https://codereview.webrtc.org/2966023002 .
> > > > > > Cr-Commit-Position: refs/heads/master@{#18928}
> > > > > > Committed: a0349c138d
> > > > >
> > > > > TBR=magjed@webrtc.org,andersc@webrtc.org
> > > > > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > > > > BUG=webrtc:7924
> > > > > NOTRY=true
> > > > >
> > > > > Review-Url: https://codereview.webrtc.org/2975963002
> > > > > Cr-Commit-Position: refs/heads/master@{#18979}
> > > > > Committed: 1095ada7ad
> > > >
> > > > R=magjed@webrtc.org
> > > > TBR=tkchin@webrtc.org
> > > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > > NOPRESUBMIT=true
> > > > NOTREECHECKS=true
> > > > NOTRY=true
> > > > BUG=webrtc:7924
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2981583002 .
> > > > Cr-Commit-Position: refs/heads/master@{#19002}
> > > > Committed: a5f1de1e65
> > >
> > > TBR=magjed@webrtc.org,tkchin@webrtc.org,jtteh@webrtc.org,andersc@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7924
> > >
> > > Review-Url: https://codereview.webrtc.org/2979973002
> > > Cr-Commit-Position: refs/heads/master@{#19004}
> > > Committed: 81d40ee149
> >
> > TBR=magjed@webrtc.org,tkchin@webrtc.org,jtteh@webrtc.org,sprang@webrtc.org
> > BUG=webrtc:7924
> >
> > Review-Url: https://codereview.webrtc.org/2979983002
> > Cr-Commit-Position: refs/heads/master@{#19005}
> > Committed: 732a3437da
>
> TBR=magjed@webrtc.org,tkchin@webrtc.org,sprang@webrtc.org,haysc@webrtc.org,andersc@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7924
>
> Review-Url: https://codereview.webrtc.org/2980173002
> Cr-Commit-Position: refs/heads/master@{#19036}
> Committed: 860f729816

TBR=magjed@webrtc.org,tkchin@webrtc.org,sprang@webrtc.org,haysc@webrtc.org,andersc@webrtc.org,jtteh@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7924

Review-Url: https://codereview.webrtc.org/2977213002
Cr-Commit-Position: refs/heads/master@{#19135}
2017-07-25 14:55:58 +00:00
danilchap
83377270dc Remove deprecated RtpRtcp::SetAudioPacketSize
was deprecated in https://codereview.webrtc.org/2545753002

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2986793002
Cr-Commit-Position: refs/heads/master@{#19134}
2017-07-25 14:46:54 +00:00
ilnik
e264a9ee9c Rename isolated_output to test_output and add a method to get the test_output directory.
BUG=none

TBR=kjellander@webrtc.org

patch from issue 2990533002 at patchset 1 (http://crrev.com/2990533002#ps1)
Already lgtm'ed in original issue. This only fixes trivial compilation errors.

Review-Url: https://codereview.webrtc.org/2989613002
Cr-Commit-Position: refs/heads/master@{#19133}
2017-07-25 14:31:18 +00:00
eladalon
42f44f9cf6 Get rid of unnecessary cast of FlexfecReceiveStreamImpl to FlexfecReceiveStream
BUG=None

Review-Url: https://codereview.webrtc.org/2967913002
Cr-Commit-Position: refs/heads/master@{#19131}
2017-07-25 13:40:06 +00:00
ilnik
59cac99c9a Report minimum PSNR in VideoQualityTest and save corresponding frame to file
BUG=none

Review-Url: https://codereview.webrtc.org/2976373002
Cr-Commit-Position: refs/heads/master@{#19130}
2017-07-25 12:45:03 +00:00
danilchap
d3f3c3497b Remove NullObjectReceiveStatistics() in rtp_rtcp module
use (already supported) nullptr as indication for no statistics

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2983363002
Cr-Commit-Position: refs/heads/master@{#19129}
2017-07-25 11:20:12 +00:00
danilchap
a04d9c31a0 Remove RtpRtcp::RemoteRTCPStat(RTCPSenderInfo*) as unused
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2986543002
Cr-Commit-Position: refs/heads/master@{#19128}
2017-07-25 11:03:39 +00:00
oprypin
d0727bfd45 Fix NSInteger formatting warning from clang 6
"error: values of type 'NSInteger' should not be used as format arguments; add an explicit cast to 'long' instead"
Casting to long is already a common practice in the code base.

This has been blocking the Chromium roll which contains an update to clang 6.0.0

BUG=None

Review-Url: https://codereview.webrtc.org/2987693002
Cr-Commit-Position: refs/heads/master@{#19127}
2017-07-25 09:04:58 +00:00
korniltsev.anatoly
ec390b5dfb When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded
There is an inconsistency in behavior of PeerConnection.
When I remove track from PeerConnection observer->OnRenegotiationNeeded is called, however if I remove track from MediaStream then there is no notification to renegotiate.
This patch adds missing OnRenegotiationNeeded calls.

BUG=webrtc:7966

Review-Url: https://codereview.webrtc.org/2977493002
Cr-Commit-Position: refs/heads/master@{#19125}
2017-07-25 00:00:25 +00:00
ehmaldonado
d083e851f6 Remove traces from {send,receive}_statistics_proxy.cc
These traces will be traced instead when getStats()
is called by JavaScript.

BUG=chromium:653087

Review-Url: https://codereview.webrtc.org/2972393002
Cr-Commit-Position: refs/heads/master@{#19124}
2017-07-24 16:00:13 +00:00
philipel
65e1f9476a Throttle log message in FrameBuffer.
BUG=webrtc:7551

Review-Url: https://codereview.webrtc.org/2987673002
Cr-Commit-Position: refs/heads/master@{#19123}
2017-07-24 15:26:53 +00:00
danilchap
c43d565873 Remove setting configuration parameter to itself.
when creating RtpRtcp module for video send stream.

BUG=webrtc:8016

Review-Url: https://codereview.webrtc.org/2979363002
Cr-Commit-Position: refs/heads/master@{#19122}
2017-07-24 15:13:34 +00:00
magjed
cc8b906467 iOS AppRTCMobile: Close peerconnection when disconnecting
We currently don't close the peerconnection before deallocing. That
could potentially cause race conditions if it's still being processed on
other threads.

BUG=webrtc:7976

Review-Url: https://codereview.webrtc.org/2976983002
Cr-Commit-Position: refs/heads/master@{#19121}
2017-07-24 14:32:33 +00:00