10800 Commits

Author SHA1 Message Date
mbonadei
4fa8a97cc3 Adding backward compatibility header
This header will be removed ad soon as downstream projects will be
updated.

BUG=webrtc:4867
NOTRY=True
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2909923002
Cr-Commit-Position: refs/heads/master@{#18306}
2017-05-29 15:20:58 +00:00
nisse
30e8931ea7 Delete RtpData::OnRecoveredPacket, use RecoveredPacketReceiver instead.
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2886813002
Cr-Commit-Position: refs/heads/master@{#18305}
2017-05-29 15:16:37 +00:00
kthelgason
580c3522d2 Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2893593002/ )
Reason for revert:
Take two of fixing downstream issues?

Original issue's description:
> Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )
>
> Reason for revert:
> Still problems with downstream projects
>
> Original issue's description:
> > Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890513002/ )
> >
> > Reason for revert:
> > Fixing downstream breakages
> >
> > Original issue's description:
> > > Revert of Split iOS sdk in to separate targets (patchset #13 id:280001 of https://codereview.webrtc.org/2862543002/ )
> > >
> > > Reason for revert:
> > > Breaking downstream projects.
> > >
> > > Original issue's description:
> > > > Split iOS sdk in to separate targets
> > > >
> > > > This CL splits the iOS sdk into separate static libraries for video,
> > > > audio, ui, common, and peerconnection-related code. This will in the
> > > > future make it easier to compile WebRTC without unneeded components.
> > > >
> > > > BUG=webrtc:4867
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2862543002
> > > > Cr-Commit-Position: refs/heads/master@{#18166}
> > > > Committed: 52c83fe710
> > >
> > > TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,kthelgason@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:4867
> > >
> > > Review-Url: https://codereview.webrtc.org/2890513002
> > > Cr-Commit-Position: refs/heads/master@{#18170}
> > > Committed: 9756238084
> >
> > TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,charujain@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:4867
> >
> > Review-Url: https://codereview.webrtc.org/2890733003
> > Cr-Commit-Position: refs/heads/master@{#18174}
> > Committed: d51e042492
>
> TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,charujain@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4867
>
> Review-Url: https://codereview.webrtc.org/2893593002
> Cr-Commit-Position: refs/heads/master@{#18182}
> Committed: 37144b214e

TBR=magjed@webrtc.org,denicija@webrtc.org,tkchin@webrtc.org,henrika@webrtc.org,charujain@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4867

Review-Url: https://codereview.webrtc.org/2893843003
Cr-Commit-Position: refs/heads/master@{#18303}
2017-05-29 12:46:00 +00:00
aleloi
048cbdda0d Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2910633002/ )
Reason for revert:
Revert of revert of revert of revert of 'Activating..'. Or "reland of reland of 'Activate..'".

*Now* the internal projects are fixed and the fix is verified.

Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ )
>
> Reason for revert:
> Reverting again: internal project issues were apparently not completely fixed.
>
> Original issue's description:
> > Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
> >
> > Reason for revert:
> > Revert the revert now that internal projects are updated.
> >
> > Original issue's description:
> > > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> > >
> > > Reason for revert:
> > > Breaks internal project.
> > >
> > > Original issue's description:
> > > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > > >
> > > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > > start/stop debug calls make file logging happen on the task queue.
> > > >
> > > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > > so that it can be shared for low priority tasks between different
> > > > subcomponents. It will require some changes to MediaEngine,
> > > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > > projects.
> > > >
> > > > A task queue must be created and destroyed from the same thread. With
> > > > this CL that will be the worker thread, which creates and destroys
> > > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > > the signaling thread.
> > > >
> > > > NOTRY=True # tests just passed
> > > >
> > > > BUG=webrtc:7404
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2896813002
> > > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > > Committed: c61bf947b4
> > >
> > > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2904893002
> > > Cr-Commit-Position: refs/heads/master@{#18255}
> > > Committed: be68b72cfa
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2903153005
> > Cr-Commit-Position: refs/heads/master@{#18270}
> > Committed: d2303a2338
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2910633002
> Cr-Commit-Position: refs/heads/master@{#18272}
> Committed: fe9ecb07ea

TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2904423002
Cr-Commit-Position: refs/heads/master@{#18300}
2017-05-29 09:56:27 +00:00
perkj
33bb86d5a1 Remove final from RtcEventLogNullImpl
The reason is that there might be implementations that do not want to implement all methods.
To allow easier modification of the RtcEventLog interface, allow these implementation to inherit the RtcEventLogNullImpl implementation.

BUG=none

Review-Url: https://codereview.webrtc.org/2903003002
Cr-Commit-Position: refs/heads/master@{#18298}
2017-05-29 09:46:05 +00:00
cstanfill
c66f8d7d6d Prevent data race in GetStaticInstance
The previous code attempted to lock instance_count and instance with a
CriticalSection, but the CriticalSection was not static, so each
function invocation got its own instance. Locking this call-specific
instance doesn't actually stop any other threads from concurrently
accessing the same function-scope globals, so this function had a data
race, which broke tsan tests (and possibly other things).

Making the CriticalSection shared among function calls will actually
synchronize access to the globals and allow our tsan tests to pass.

BUG=webrtc:3062

Review-Url: https://codereview.webrtc.org/2890213002
Cr-Commit-Position: refs/heads/master@{#18296}
2017-05-29 07:01:14 +00:00
deadbeef
b56671e051 Fix issue with send-side bandwidth estimation over TURN TCP connections.
AsyncStunTCPSocket wasn't firing SignalSentPacket, which the bandwidth
estimator requires for every packet in order to look up send times when
feedback arrives. If the signal isn't fired, it always assumes feedback
is arriving extremely late, and decreases the bandwidth by a factor of
2 until it reaches the minimum of 10kbps.

BUG=webrtc:7717
TBR=pthatcher@webrtc.org

Review-Url: https://codereview.webrtc.org/2912523003
Cr-Commit-Position: refs/heads/master@{#18279}
2017-05-27 01:40:05 +00:00
deadbeef
eae4564cb7 Disable SIGPIPE for sockets created on iOS.
This can occur (and by default, terminates the process) for apps that
don't use the "voip" UIBackgroundMode.

We're already doing a similar thing on Linux (using MSG_NOSIGNAL for every
packet sent).

BUG=webrtc:7686

Review-Url: https://codereview.webrtc.org/2903313002
Cr-Commit-Position: refs/heads/master@{#18277}
2017-05-26 23:27:09 +00:00
aleloi
fe9ecb07ea Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2903153005/ )
Reason for revert:
Reverting again: internal project issues were apparently not completely fixed.

Original issue's description:
> Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
>
> Reason for revert:
> Revert the revert now that internal projects are updated.
>
> Original issue's description:
> > Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
> >
> > Reason for revert:
> > Breaks internal project.
> >
> > Original issue's description:
> > > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> > >
> > > A low priority task queue is added to WebRTCVoiceEngine. The
> > > start/stop debug calls make file logging happen on the task queue.
> > >
> > > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > > so that it can be shared for low priority tasks between different
> > > subcomponents. It will require some changes to MediaEngine,
> > > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > > projects.
> > >
> > > A task queue must be created and destroyed from the same thread. With
> > > this CL that will be the worker thread, which creates and destroys
> > > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > > the signaling thread.
> > >
> > > NOTRY=True # tests just passed
> > >
> > > BUG=webrtc:7404
> > >
> > > Review-Url: https://codereview.webrtc.org/2896813002
> > > Cr-Commit-Position: refs/heads/master@{#18252}
> > > Committed: c61bf947b4
> >
> > TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2904893002
> > Cr-Commit-Position: refs/heads/master@{#18255}
> > Committed: be68b72cfa
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2903153005
> Cr-Commit-Position: refs/heads/master@{#18270}
> Committed: d2303a2338

TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2910633002
Cr-Commit-Position: refs/heads/master@{#18272}
2017-05-26 12:46:34 +00:00
aleloi
d2303a2338 Reland of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #1 id:1 of https://codereview.webrtc.org/2904893002/ )
Reason for revert:
Revert the revert now that internal projects are updated.

Original issue's description:
> Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
>
> Reason for revert:
> Breaks internal project.
>
> Original issue's description:
> > Activate 'offload debug dump recordings from audio thread to TaskQueue'.
> >
> > A low priority task queue is added to WebRTCVoiceEngine. The
> > start/stop debug calls make file logging happen on the task queue.
> >
> > In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> > so that it can be shared for low priority tasks between different
> > subcomponents. It will require some changes to MediaEngine,
> > CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> > projects.
> >
> > A task queue must be created and destroyed from the same thread. With
> > this CL that will be the worker thread, which creates and destroys
> > WebRTCVoiceEngine. With the dependent CL, it will probably change to
> > the signaling thread.
> >
> > NOTRY=True # tests just passed
> >
> > BUG=webrtc:7404
> >
> > Review-Url: https://codereview.webrtc.org/2896813002
> > Cr-Commit-Position: refs/heads/master@{#18252}
> > Committed: c61bf947b4
>
> TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2904893002
> Cr-Commit-Position: refs/heads/master@{#18255}
> Committed: be68b72cfa

TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2903153005
Cr-Commit-Position: refs/heads/master@{#18270}
2017-05-26 12:13:18 +00:00
sakal
407e3afd37 Add sakal@webrtc.org as an owner of examples/androidtests.
BUG=None
NOTRY=True

Review-Url: https://codereview.webrtc.org/2886363004
Cr-Commit-Position: refs/heads/master@{#18269}
2017-05-26 10:24:12 +00:00
sakal
d7fdb8014d Reland of Removes usage of native base::android::GetApplicationContext()
The change is now compatible with the old JVM::Initialize API. The
context is passed to the ContextUtils class when calling its deprecated
signature.

BUG=webrtc:7665
NOTRY=True # Only comment changes since the last patchset.

Review-Url: https://codereview.webrtc.org/2903253004
Cr-Commit-Position: refs/heads/master@{#18268}
2017-05-26 08:51:53 +00:00
jtteh
13ae11a418 Add observer for AVAudioSession.outputVolume
BUG=webrtc:7696

Review-Url: https://codereview.webrtc.org/2895263006
Cr-Commit-Position: refs/heads/master@{#18267}
2017-05-26 00:52:20 +00:00
deadbeef
a615e17ec0 Allow constructing an EglBase from an existing shared EGLContext.
BUG=None

Review-Url: https://codereview.webrtc.org/2885163003
Cr-Commit-Position: refs/heads/master@{#18266}
2017-05-25 17:11:25 +00:00
deadbeef
8b7e9ad554 Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
This CL doesn't yet offer these protos; it just accepts them if they're
seen in a remote offer. It also doesn't verify that the ICE candidate
protocol matches the m= section protocol (UDP vs. TCP), since we don't
do this elsewhere and don't really have a reason to care.

This CL also adds an integration test that receives a spec-compliant
SCTP offer and attempts to send data bidirectionally.

BUG=webrtc:7706

Review-Url: https://codereview.webrtc.org/2902213002
Cr-Commit-Position: refs/heads/master@{#18265}
2017-05-25 16:38:55 +00:00
eladalon
edd6eea542 Rename elad.alon to eladalon, to avoid confusion between repositories.
BUG=None
NOTRY=true

Review-Url: https://codereview.webrtc.org/2899303002
Cr-Commit-Position: refs/heads/master@{#18264}
2017-05-25 07:15:35 +00:00
lliuu
548cdce7bc Revert of https://codereview.webrtc.org/2889183002/
And also revert https://codereview.webrtc.org/2888093005/ (Chromium roll) which has a dependency on 2889183002

BUG=webrtc:7707

Review-Url: https://codereview.webrtc.org/2897423002
Cr-Commit-Position: refs/heads/master@{#18263}
2017-05-24 23:45:57 +00:00
jianj
20acdf2443 Add vp9 QP parser.
BUG=webrtc:7662

Review-Url: https://codereview.webrtc.org/2891803003
Cr-Commit-Position: refs/heads/master@{#18260}
2017-05-24 17:00:16 +00:00
eladalon
ae550e397a Correct sequence-number injection into packets in rtp_packet_unittest.cc
BUG=None

Review-Url: https://codereview.webrtc.org/2899293002
Cr-Commit-Position: refs/heads/master@{#18257}
2017-05-24 15:28:13 +00:00
aleloi
be68b72cfa Revert of Activate 'offload debug dump recordings from audio thread to TaskQueue'. (patchset #4 id:160001 of https://codereview.webrtc.org/2896813002/ )
Reason for revert:
Breaks internal project.

Original issue's description:
> Activate 'offload debug dump recordings from audio thread to TaskQueue'.
>
> A low priority task queue is added to WebRTCVoiceEngine. The
> start/stop debug calls make file logging happen on the task queue.
>
> In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
> so that it can be shared for low priority tasks between different
> subcomponents. It will require some changes to MediaEngine,
> CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
> projects.
>
> A task queue must be created and destroyed from the same thread. With
> this CL that will be the worker thread, which creates and destroys
> WebRTCVoiceEngine. With the dependent CL, it will probably change to
> the signaling thread.
>
> NOTRY=True # tests just passed
>
> BUG=webrtc:7404
>
> Review-Url: https://codereview.webrtc.org/2896813002
> Cr-Commit-Position: refs/heads/master@{#18252}
> Committed: c61bf947b4

TBR=solenberg@webrtc.org,tommi@webrtc.org,perkj@webrtc.org,danilchap@webrtc.org,tommi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2904893002
Cr-Commit-Position: refs/heads/master@{#18255}
2017-05-24 11:19:54 +00:00
kwiberg
0afb13014b AudioEncoderPcm16B: Number of bits/sample is 16, not 2
Clearly, this was a case of bit/byte confusion.

BUG=none

Review-Url: https://codereview.webrtc.org/2904883002
Cr-Commit-Position: refs/heads/master@{#18254}
2017-05-24 11:09:52 +00:00
aleloi
c61bf947b4 Activate 'offload debug dump recordings from audio thread to TaskQueue'.
A low priority task queue is added to WebRTCVoiceEngine. The
start/stop debug calls make file logging happen on the task queue.

In a dependent CL (https://codereview.webrtc.org/2888303003), the task queue is moved to PeerConnectionFactory,
so that it can be shared for low priority tasks between different
subcomponents. It will require some changes to MediaEngine,
CompositeMediaEngine, WebRTCVoiceEngine, and changes in internal
projects.

A task queue must be created and destroyed from the same thread. With
this CL that will be the worker thread, which creates and destroys
WebRTCVoiceEngine. With the dependent CL, it will probably change to
the signaling thread.

NOTRY=True # tests just passed

BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2896813002
Cr-Commit-Position: refs/heads/master@{#18252}
2017-05-24 08:47:18 +00:00
elad.alon
3696f7efc4 Remove unneeded gmock header from RPLR UT
BUG=None

Review-Url: https://codereview.webrtc.org/2898193002
Cr-Commit-Position: refs/heads/master@{#18250}
2017-05-24 07:13:25 +00:00
peah
1b92722ad5 Simplified the ERLE computation code in AEC3
BUG=webrtc:7519

Review-Url: https://codereview.webrtc.org/2901253002
Cr-Commit-Position: refs/heads/master@{#18249}
2017-05-24 06:44:47 +00:00
deadbeef
aea9293fd4 Revert of Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations. (patchset #3 id:40001 of https://codereview.webrtc.org/2885143006/ )
Reason for revert:
Causes a new TSan race warning. Will reland after fixing. Note this is the same race as will be fixed by https://codereview.webrtc.org/2876273002/.

Original issue's description:
> Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.
>
> The deadlock occurs if the AsyncInvoker is destroyed on thread A while
> a task on thread B is running, which AsyncInvokes a task back on thread
> A.
>
> This was causing pending_invocations_ to end up negative, because
> an AsyncClosure that's never added to a thread's message queue (due to
> the "destroying_" flag) caused the count to be decremented but not
> incremented.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2885143006
> Cr-Commit-Position: refs/heads/master@{#18225}
> Committed: ef37ca5fb3

TBR=nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2904543003
Cr-Commit-Position: refs/heads/master@{#18248}
2017-05-23 19:55:03 +00:00
danilchap
b34f6a8520 Remove deprecated isFirstPacket member name
Original rename .isFirstPacket to .is_first_packet_in_frame
was done in https://codereview.webrtc.org/2614503002

BUG=None

Review-Url: https://codereview.webrtc.org/2895473004
Cr-Commit-Position: refs/heads/master@{#18247}
2017-05-23 18:06:56 +00:00
stefan
95e9754e40 More gracefully handle timing errors, such as unexpected changes in the rtp timestamp.
BUG=webrtc:7682

Review-Url: https://codereview.webrtc.org/2898763005
Cr-Commit-Position: refs/heads/master@{#18245}
2017-05-23 16:52:18 +00:00
ilnik
7a3006bae7 Fix packetization logic to leave space for extensions in the last packet
Change packetizer interface to explicitly return number of packets
instead of a last flag. Account for extra space needed in the last
packet.

BUG=webrtc:7588,webrtc:7594

Review-Url: https://codereview.webrtc.org/2871173008
Cr-Commit-Position: refs/heads/master@{#18244}
2017-05-23 16:34:21 +00:00
braveyao
d019667c00 Linux desktopCapture: fix the cursor position issue in Window sharing
On Linux, during Windwo sharing, the cursore capture may happen in the parent
window of the target. And the parent window may have some decorations added by
window manager(Chrome windows don't have those decorations.), so the relative
cursor position to the parent window with decorations may differ to its child
target window. The offset includes the height of caption bar and the around
shadow and border.
This problem only happens with Window sharing on Linux.

The fix is to translate the coordinates from the parent window to the coordinates space of the target window.

BUG=723889

Review-Url: https://codereview.webrtc.org/2889063002
Cr-Commit-Position: refs/heads/master@{#18243}
2017-05-23 16:31:14 +00:00
ehmaldonado
c1b5ea959e Add traces for some video receive statistics.
This CL adds traces to compute the following metrics that getStats()
captures for video:
- googFrameRateOutput
- packetsLost
- googFrameWidthReceived
- googFrameHeightReceived
- googCurrentDelayMs
- googTargetDelayMs
- googDecodeMs
- googMaxDecodeMs
- googJitterBufferMs
- googRenderDelayMs

BUG=chromium:653087

Review-Url: https://codereview.webrtc.org/2884643004
Cr-Commit-Position: refs/heads/master@{#18242}
2017-05-23 16:06:13 +00:00
aleloi
06013e99ac AecDump implementation.
This CL implements webrtc::AecDump, which is an interface defined in
https://codereview.webrtc.org/2778783002.

This AudioProcessing submodule writes audio and APM state to a
file. The file writing is done by posting IO tasks
(write_to_file_task.h) on an rtc::TaskQueue. There is an existing
implementation for this through AudioProcessing::StartDebugRecording()
and AudioProcessing::StopDebugRecording(). This implementation still
works, and is used as the default until this dependent CL:
https://codereview.webrtc.org/2896813002/.

To be able to build webrtc without protobuf support, the interface is
isolated from protobuf types. Audio data from AudioProcessing is
passed to AecDumpImpl through the AecDump interface. There it is
stored in protobuf objects, which are posted on the task queue.

This functionality is verified correct by the CL
https://codereview.webrtc.org/2864373002, which enables this recording
submodule in APM tests.

BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2865113002
Cr-Commit-Position: refs/heads/master@{#18241}
2017-05-23 15:52:05 +00:00
brandtr
f27c5b8d6e Add FlexfecReceiver unit test for infinite recovery loop.
This CL adds unit tests to the FlexfecReceiver, verifying that the
infinite recovery loop described in
https://codereview.webrtc.org/2867943003/ is tested for.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2895083002
Cr-Commit-Position: refs/heads/master@{#18240}
2017-05-23 15:38:43 +00:00
jansson
cc8b14b1ec Revert of Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric n… (patchset #1 id:1 of https://codereview.webrtc.org/2901793002/ )
Reason for revert:
Looks like we need the return there after all: derefence error in: https://luci-logdog.appspot.com/v/?s=chromium%2Fbb%2Fclient.webrtc.perf%2FAndroid64_Tests__L_Nexus9_%2F2967%2F%2B%2Frecipes%2Fsteps%2Fvideo_quality_loopback_test%2F0%2Fstdout

Original issue's description:
> Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric numbered list
>
> BUG=webrtc:7698
> NOTRY=TRUE
>
> Review-Url: https://codereview.webrtc.org/2901793002
> Cr-Commit-Position: refs/heads/master@{#18234}
> Committed: 18d023f9ee

TBR=kjellander@webrtc.org,oprypin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7698

Review-Url: https://codereview.webrtc.org/2899973004
Cr-Commit-Position: refs/heads/master@{#18239}
2017-05-23 15:31:11 +00:00
philipel
7d79e63a48 Cast sequence number in RtpFrameObject.
BUG=webrtc:7700

Review-Url: https://codereview.webrtc.org/2902743002
Cr-Commit-Position: refs/heads/master@{#18237}
2017-05-23 15:19:11 +00:00
sprang
916170ae46 Don't boost QP after drop unless there is sufficient bandwidth
If a frame is dropped and re-encoded because it exceeded the target
bitrate by a large factor, the next frame will be encoded at max qp
(worst quality) in order to get a frame through in a timely manner. The
next frame after this will still have lower quality since the rate
controller essentially gets reset. In order to mitigate that we boost
the qp for that next frame, which brings the stream back to a good
quality quicker.

However, if the network conditions are _really_ bad, this boosted qp
may be too large, causing the frame again to be dropped an re-encoded.

This CL set's a minimum bitrate available in order to enabling the
boosting in the first place.
It also adjusts a timeout (max time between frames in TL0), since a
too small value and very difficult frames in conjunction with the
mentioned bad network could actually cause bad network over-utilization
in turn leading to packet loss and bad follow-on effects to that.

There was also some slop in the rate keeping for the two layers.
This has been tightened up and affected test cases have been fixed.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2897983002
Cr-Commit-Position: refs/heads/master@{#18236}
2017-05-23 14:47:55 +00:00
sakal
7855fff5bf Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ )
Reason for revert:
Fix issue.

Original issue's description:
> Revert of Removes usage of native base::android::GetApplicationContext() (patchset #6 id:120001 of https://codereview.webrtc.org/2888093004/ )
>
> Reason for revert:
> Breaks bot on chromium.webrtc.fyi.
>
> Original issue's description:
> > Removes usage of native base::android::GetApplicationContext()
> >
> > BUG=webrtc:7665
> >
> > Review-Url: https://codereview.webrtc.org/2888093004
> > Cr-Commit-Position: refs/heads/master@{#18195}
> > Committed: bc83e2ee69
>
> TBR=magjed@webrtc.org,henrika@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7665
>
> Review-Url: https://codereview.webrtc.org/2894593002
> Cr-Commit-Position: refs/heads/master@{#18196}
> Committed: 40d224814a

BUG=webrtc:7665

Review-Url: https://codereview.webrtc.org/2889183002
Cr-Commit-Position: refs/heads/master@{#18235}
2017-05-23 14:34:17 +00:00
jansson
18d023f9ee Remove unecessary non fatal error statement that very often is printed in the PSNR or SSIM metric numbered list
BUG=webrtc:7698
NOTRY=TRUE

Review-Url: https://codereview.webrtc.org/2901793002
Cr-Commit-Position: refs/heads/master@{#18234}
2017-05-23 14:26:41 +00:00
aleloi
868f32f423 AudioProcessingModule has a feature to make a recording of its
configuration, inputs and outputs over a period of time. It is
activated by AudioProcessing::StartRecording. The data is stored in
binary protobuf format in a specified file. The file IO is, as of
this CL, done from the real-time audio thread.

This CL contains an interface for AecDump, a new APM submodule that
will handle the recordings. Calls to the new interface from the
AudioProcessingModule are added. These calls have no effect, and for a
short while, audio_processing_impl.cc will contain two copies of
recording calls.

The original calls are guarded by the WEBRTC_AUDIOPROC_DEBUG_DUMP
preprocessor define. They still have an effect, while the new ones do
not. In the following CLs, the old recording calls will be removed,
and an implementation of AecDump added.

The reasons for the refactoring is to move file IO operations from the
real-time audio thread, to add a top-level low-priority task queue for
logging tasks like this, to simplify and modularize audio_processing_impl.cc
and remove some of the preprocessor directives. These goals will be
archived by the upcoming CLs. The implementation is in
https://codereview.webrtc.org/2865113002.

BUG=webrtc:7404

Review-Url: https://codereview.webrtc.org/2778783002
Cr-Commit-Position: refs/heads/master@{#18233}
2017-05-23 14:20:05 +00:00
sprang
dceb42da3e Update screen simulcast config and fix periodic encoder param update
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.

Also fix a bug in vie_encoder where the codec was not perioducally
updated unless a new bitrate estimate was triggered.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
2017-05-23 13:45:08 +00:00
ossu
c3d4b48e7e Store/restore RTP state for audio streams with same SSRC within a call
This functionality already exists for video streams, so not having it
for audio is unexpected and has lead to problems.

BUG=webrtc:7631

Review-Url: https://codereview.webrtc.org/2887733002
Cr-Commit-Position: refs/heads/master@{#18231}
2017-05-23 13:07:11 +00:00
peah
23ac8b49f4 Preserve level controller output when no other effects are active
This CL ensures that the output of the level controller is kept
when no other submodules in APM are active

BUG=webrtc:7697,

Review-Url: https://codereview.webrtc.org/2902723002
Cr-Commit-Position: refs/heads/master@{#18230}
2017-05-23 12:33:56 +00:00
peah
1d68089f4b Transparency increasing tuning for AEC3.
This CL increases the transparency of the AEC3 via tuning.
The major changes are
1) Limiting the suppression gain to the 16 bit sample floor.
2) Controlling the rate of the suppression gain increase
   according to the signal characteristics.

Apart from these tunings, the code for the suppression gain
was refactored to increase/maintain the code quality after
the above changes.

BUG=webrtc:7519,webrtc:7528, chromium:715893

Review-Url: https://codereview.webrtc.org/2886733002
Cr-Commit-Position: refs/heads/master@{#18229}
2017-05-23 11:07:10 +00:00
brandtr
5e171752a2 Reland of use allocated encoders in SimulcastEncoderAdapter. (patchset #1 id:1 of https://codereview.webrtc.org/2893003002/ )
Reason for reland:
Chrome encoder implementation fixed.

Original issue's description:
> Revert of Reuse allocated encoders in SimulcastEncoderAdapter. (patchset #15 id:320001 of https://codereview.webrtc.org/2830793005/ )
>
> Reason for revert:
> Breaks Chrome tests.
>
> Original issue's description:
> > Reuse allocated encoders in SimulcastEncoderAdapter.
> >
> > Prior to this change, the SimulcastEncoderAdapter would destroy and create
> > encoders whenever it is being reinitialized. After this change, the
> > SimulcastEncoderAdapter will cache the already allocated encoders, and reuse
> > them after reinitialization.
> >
> > This change will help in reducing the number of PictureID "jumps" that have
> > been seen around encoder reinitialization.
> >
> > TESTED=AppRTCMobile, Chrome desktop, and internal app, with forced encoder reinits every 30 frames and https://codereview.webrtc.org/2833493003/ applied.
> > BUG=webrtc:7475
> >
> > Review-Url: https://codereview.webrtc.org/2830793005
> > Cr-Commit-Position: refs/heads/master@{#18215}
> > Committed: 0b8bfb9d98
>
> TBR=stefan@webrtc.org,noahric@chromium.org,glaznev@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7475
>
> Review-Url: https://codereview.webrtc.org/2893003002
> Cr-Commit-Position: refs/heads/master@{#18216}
> Committed: 56e119e2e8

TBR=stefan@webrtc.org,noahric@chromium.org,glaznev@webrtc.org,sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7475

Review-Url: https://codereview.webrtc.org/2901493002
Cr-Commit-Position: refs/heads/master@{#18228}
2017-05-23 10:32:16 +00:00
peah
8a8ebd94b0 Field trial support to whenever possible turn off the AGC and HPF
When operating on mobile devices, where hardware support is available
for the AEC and NS functionality, it is desirable to be able to
operate without hardcoded behaviors for the WebRTC AGC and HPF.

This CL adds support to allow a field trial to turn these off
whenever that is possible.

BUG=webrtc:6220, webrtc:6183, webrtc:6181

Review-Url: https://codereview.webrtc.org/2876133002
Cr-Commit-Position: refs/heads/master@{#18226}
2017-05-22 22:48:47 +00:00
deadbeef
ef37ca5fb3 Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.
The deadlock occurs if the AsyncInvoker is destroyed on thread A while
a task on thread B is running, which AsyncInvokes a task back on thread
A.

This was causing pending_invocations_ to end up negative, because
an AsyncClosure that's never added to a thread's message queue (due to
the "destroying_" flag) caused the count to be decremented but not
incremented.

BUG=None

Review-Url: https://codereview.webrtc.org/2885143006
Cr-Commit-Position: refs/heads/master@{#18225}
2017-05-22 22:32:51 +00:00
perkj
f472699bbd Replace AudioSendStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2856063003
Cr-Commit-Position: refs/heads/master@{#18224}
2017-05-22 17:12:26 +00:00
perkj
ac8f52de70 Replace AudioReceiveStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2851303007
Cr-Commit-Position: refs/heads/master@{#18223}
2017-05-22 16:36:28 +00:00
alessiob
3ec96df907 This CL introduces a new APM sub-module named AGC2 that does not use the band
split domain and only implements floating point operations (to avoid spectral
leakage issues and unnecessary complexity).

The goal of this CL is adding the new sub-module into APM without providing an
implementation that could replace the existing gain control modules. The focus
is in fact on initialization, reset, and configuration of AGC2.

The module itself only applies a hard-coded gain value. This behavior will
change in the coming CLs.

BUG=webrtc:7494

Review-Url: https://codereview.webrtc.org/2848593002
Cr-Commit-Position: refs/heads/master@{#18222}
2017-05-22 13:57:06 +00:00
perkj
c0876aab46 Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2857933002
Cr-Commit-Position: refs/heads/master@{#18221}
2017-05-22 11:08:28 +00:00
perkj
09e71daec5 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2850793002
Cr-Commit-Position: refs/heads/master@{#18220}
2017-05-22 10:26:49 +00:00