8673 Commits

Author SHA1 Message Date
henrik.lundin
834a6ea12b Add muted_output parameter to ACM
The new parameter indicates if the output in the AudioFrame is muted. If
so, the output samples are not written, but should be interpreted as all
zero.

A version of AudioCodingModule::PlayoutData10Ms() without the new
parameter is maintained while waiting for downstream dependencies to
conform.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1976913002
Cr-Commit-Position: refs/heads/master@{#12719}
2016-05-13 10:45:31 +00:00
philipel
29dca2ce95 Added cluster id to PacedSender::Callback::TimeToSendPacket.
Also added cluster id to paced_sender::Packet and set the cluster id of
the probing packet that is about to be sent.

BUG=webrtc:5859
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1962303002 .

Cr-Commit-Position: refs/heads/master@{#12718}
2016-05-13 09:13:16 +00:00
philipel
1a830c2c66 Nack count returned on OnReceivedPacket.
OnReceivedPacket now return the number of times the packet has been nacked. Also some minor refactoring.

BUG=webrtc:5514
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1972123002 .

Cr-Commit-Position: refs/heads/master@{#12717}
2016-05-13 09:12:11 +00:00
mflodman
2ebe5b1cd8 Refactor before implementing per stream suspension.
This CL contains a few minor changes to names, function signatures and
merges two structs into one.

BUG=5868

Review-Url: https://codereview.webrtc.org/1952923005
Cr-Commit-Position: refs/heads/master@{#12716}
2016-05-13 08:43:56 +00:00
perkj
7339c500fe Revert of Remove ViEEncoder::SetNetworkStatus (patchset #11 id:200001 of https://codereview.webrtc.org/1932683002/ )
Reason for revert:
Breaks Chrome FYI using H264.
Need to investigate.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4170

Original issue's description:
> Remove ViEEncoder::SetNetworkStatus
>
> This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
>
> BUG=webrtc:5687
> NOTRY=True
>
> Committed: https://crrev.com/50b5c3be844ef571a28b2681c549443a26735d72
> Cr-Commit-Position: refs/heads/master@{#12699}

TBR=stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1978783002
Cr-Commit-Position: refs/heads/master@{#12715}
2016-05-13 08:17:37 +00:00
terelius
d5c1a0bd5d New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs.
Changed rtpdump converter and neteq tool to use new parser, but still aborting if the file is corrupt.

Review-Url: https://codereview.webrtc.org/1768773002
Cr-Commit-Position: refs/heads/master@{#12714}
2016-05-13 07:43:04 +00:00
peah
5df729489f Refactored the comfort noise generation code in the AEC.
This CL will be followed with other CLs that break apart
the application of the comfort noise from the comfort
noise generation.

The changes in the CL are very close to bitexaxt. The
bitinexactness is caused by differences in numerical
behavior when bundling the spectral band power and the
noise scaling based on the NLP gain.

BUG=webrtc:5201, webrtc:5298

Review-Url: https://codereview.webrtc.org/1958933002
Cr-Commit-Position: refs/heads/master@{#12713}
2016-05-13 07:13:57 +00:00
peah
9bbf89bca1 Moved the AEC echo suppression gain computation code to
a separate method.

This CL will be followed by other CLs that simplify this method and break out the state specific to this computation
into a separate substate.

The changes are bitexact.

BUG=webrtc:5201, webrtc:5298

Review-Url: https://codereview.webrtc.org/1963493003
Cr-Commit-Position: refs/heads/master@{#12712}
2016-05-13 06:08:11 +00:00
henrik.lundin
7a926812d8 NetEq: Implement muted output
This CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
2016-05-12 20:51:37 +00:00
deadbeef
c55fb30649 Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.

I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.

Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51

TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
2016-05-12 19:51:45 +00:00
Taylor Brandstetter
48e9d05f51 Implement RTCConfiguration.iceCandidatePoolSize.
It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).

This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.

R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1956453003 .

Cr-Commit-Position: refs/heads/master@{#12708}
2016-05-12 17:19:44 +00:00
Honghai Zhang
6705012904 This fixes an issue similar to
https://bugs.chromium.org/p/webrtc/issues/detail?id=3927
where the localhost IP does not match the turn port address.
The issue here is in the TCP port.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1914803002 .

Cr-Commit-Position: refs/heads/master@{#12707}
2016-05-12 16:28:08 +00:00
Taylor Brandstetter
dc4eb8c5b3 Refactoring some tests in peerconnectioninterface_unittest.cc.
Some tests were passing in a local description created from hard-coded
SDP strings, which won't work in the future (since some attributes such
as the fingerprint and ICE ufrag/pwd are non-modifiable). These tests
now do the typical approach of calling CreateOffer and modifying the
result if necessary.

Also added some non-const versions of the SessionDescription accessor
helper functions, since that makes it much easier to modify a
SessionDescription. Previous alternatives were re-implementing the
helper methods from scratch, or converting the description to SDP,
modifying it, and converting it back.

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1966333002 .

Cr-Commit-Position: refs/heads/master@{#12704}
2016-05-12 15:14:54 +00:00
Peter Boström
d8b0109327 Fix RTX-configuration test with >2 codecs built.
Fixes WebRtcVideoChannel2Test.DefaultReceiveStreamReconfiguresToUseRtx
under rtc_use_h264=1.

BUG=webrtc:5816
R=danilchap@webrtc.org

Review URL: https://codereview.webrtc.org/1938503002 .

Cr-Commit-Position: refs/heads/master@{#12703}
2016-05-12 14:44:46 +00:00
Danil Chapovalov
d215ade504 [rtcp] Remb::Parse updated not to use RTCPUtility
bitrate field changed to 64bit to match Remb packet format

BUG=webrtc:5260
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1959023002 .

Cr-Commit-Position: refs/heads/master@{#12702}
2016-05-12 13:25:50 +00:00
peah
b1fc54d33e Corrected the delay agnostic AEC behavior during periods of silent farend signal.
Added conditional updating of the statistics and the delay estimate so that
updates are only done when the farend is non-stationary.

The reason for this is that all the values that go into the updating of the
statistics, and that in turn are also used to update the delay, are frozen
when the farend signal is non-stationary. Therefore, when the farend signal
is silent (stationary), the last estimates present before the silent (stationary)
period began are used to continue to update the statistics. This is a problem as
the updating is done in a manner that assumes that the estimates continue
to be updated.

This CL conditions the updating based on stationarity instead of silence
as both are treated in the same manner in the delay agnostic AEC.
This makes sense theoretically as the delay agnostic AEC operates on
analyzing power deviations (in bands) from a slowly updated average power and
therefore for a stationary signal will have no such deviations to base its analysis
on.

BUG=webrtc:5875, chromium:576624

NOTRY=True

Review-Url: https://codereview.webrtc.org/1967033002
Cr-Commit-Position: refs/heads/master@{#12700}
2016-05-12 12:08:53 +00:00
perkj
50b5c3be84 Remove ViEEncoder::SetNetworkStatus
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.

BUG=webrtc:5687
NOTRY=True

Review-Url: https://codereview.webrtc.org/1932683002
Cr-Commit-Position: refs/heads/master@{#12699}
2016-05-12 11:53:52 +00:00
magjed
b9253060b8 Add magjed@ and perkj@ as webrtc/examples/ owners
NOTRY=true

Review-Url: https://codereview.webrtc.org/1969403002
Cr-Commit-Position: refs/heads/master@{#12698}
2016-05-12 10:48:26 +00:00
Magnus Jedvert
210dd5c361 VideoCapturerAndroid: Ignore erroneous startCaptureOnCameraThread calls instead of crashing
Fix a bug where startCaptureOnCameraThread() is called while the camera is already successfully running. It may happen in the scenario when startCapture() is called, but startCaptureOnCameraThread() fails and posts a retry, then stopCapture() is called and removeCallbacksAndMessages() fails to remove the pending retry, and then startCapture() is called successfully.

BUG=b/28181364
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1967053002 .

Cr-Commit-Position: refs/heads/master@{#12697}
2016-05-12 10:40:36 +00:00
mostynb
e38e4f6e48 IWYU: errno.h in base/logging.h
Without this, some toolchains may fail to build base/checks.cc
because errno is undefined.

NOTRY=true

Review-Url: https://codereview.webrtc.org/1971513002
Cr-Commit-Position: refs/heads/master@{#12696}
2016-05-12 08:08:29 +00:00
Magnus Jedvert
060aa57084 VideoCapturerAndroid: Force setDisplayOrientation to 0
BUG=b/27994417
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1968913002 .

Cr-Commit-Position: refs/heads/master@{#12695}
2016-05-12 08:08:00 +00:00
Danil Chapovalov
7f216b71aa Renames TransportController worker_thread to network_thread.
function suffix '_w' changes to '_n'

BUG=webrtc:5645
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1895813003 .

Cr-Commit-Position: refs/heads/master@{#12694}
2016-05-12 07:20:43 +00:00
Henrik Kjellander
3fe372dbee Fix all -Wnon-virtual-dtor warnings.
This is needed to get the GN build going for several parts
of the code tree.

BUG=webrtc:3307
NOTRY=True
R=henrika@webrtc.org, nisse@webrtc.org

Review URL: https://codereview.webrtc.org/1928653005 .

Cr-Commit-Position: refs/heads/master@{#12693}
2016-05-12 06:11:09 +00:00
Peter Boström
ad6fc5a05c Remove remaining quality-analysis (QM).
This was never turned on, contains a lot of complexity and somehow
manages triggering a bug in a downstream project.

BUG=webrtc:5066
R=marpan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1917323002 .

Cr-Commit-Position: refs/heads/master@{#12692}
2016-05-12 01:01:42 +00:00
Peter Boström
919288f6ba Clamp number of downscales in QualityScaler.
Fixes bug where QualityScaler would be stuck "way below" QVGA (due to
downscale_shift_) even though it would never scale below QVGA. Also
fixes issue where samples would be cleared when either staying at max
resolution or going below QVGA even though no action happened.

BUG=
R=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1971693003 .

Cr-Commit-Position: refs/heads/master@{#12691}
2016-05-12 00:17:52 +00:00
Danil Chapovalov
33b01f2162 Adds network thread to rtc::BaseChannel
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.

BUG=webrtc:5645
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1903393004 .

Cr-Commit-Position: refs/heads/master@{#12690}
2016-05-11 17:55:41 +00:00
Honghai Zhang
3108fc933b Add config continualGatheringPolicy to the IOS RTCConfiguration.
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1971563002 .

Cr-Commit-Position: refs/heads/master@{#12689}
2016-05-11 17:10:47 +00:00
perkj
ec81bcd519 Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate
This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/)

Original reverted cl is in patch set #1.
Changes in following patch sets.

The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)()

It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate.

Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)

BUG=chromium:609816, webrtc:5687
TBR=mflodman@webrtc.org
NOTRY=True  // Due to bug  in android_x86 cq builder....

Review-Url: https://codereview.webrtc.org/1958113003
Cr-Commit-Position: refs/heads/master@{#12688}
2016-05-11 13:01:19 +00:00
kjellander
2f5ae66471 Add root owners to webrtc/OWNERS
For WebRTC inside Chromium, only the webrtc/ directory is present
(as src/third_party/webrtc). That makes it impossible to add DEPS
check_deps rules in Chromium without approval of a webrtc.org owner
(see https://codereview.chromium.org/1818903004).

By having our root owners also be owners in webrtc/, this should be
less confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/1934523002
Cr-Commit-Position: refs/heads/master@{#12687}
2016-05-11 12:47:43 +00:00
kwiberg
6ab3db249b Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!

Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
2016-05-11 12:07:33 +00:00
ossu
7e3968e46c Removed MaxEncodedBytes from AudioEncoder.
This is the last step in changing the signature of AudioEncoder::Encode
to taking an rtc::Buffer as its output parameter, rather than a pointer
to and a size parameter.

The notry parameter has been added specifically to work around android_compile_x86_dbg bot failing.

NOTRY=True
BUG=webrtc:5591

Review-Url: https://codereview.webrtc.org/1962013003
Cr-Commit-Position: refs/heads/master@{#12685}
2016-05-11 11:39:58 +00:00
kwiberg
65fc62e9dd Remove webrtc/base/scoped_ptr.h
BUG=webrtc:5520

NOTRY=True

Review-Url: https://codereview.webrtc.org/1942823002
Cr-Commit-Position: refs/heads/master@{#12684}
2016-05-11 11:29:38 +00:00
kwiberg
8a70714851 Modernize variable names
As promised in
https://codereview.webrtc.org/1946873003/diff/1/webrtc/modules/utility/source/coder.h#newcode54

NOTRY=True

Review-Url: https://codereview.webrtc.org/1968853002
Cr-Commit-Position: refs/heads/master@{#12683}
2016-05-11 11:26:59 +00:00
Fredrik Solenberg
cd6ae6652f Removing some old code which looked like it had to do with NACK handling but in reality did nothing.
BUG=webrtc:5762, webrtc:4690
R=stefan@webrtc.org
TBR=mflodman

Review URL: https://codereview.webrtc.org/1946183002 .

Cr-Commit-Position: refs/heads/master@{#12682}
2016-05-11 11:05:13 +00:00
Henrik Boström
faa78dc38d Removed old DtlsIdentityRequestObserver::RequestIdentity function signature
since the new signature is used everywhere.

BUG=chromium:544902, webrtc:5092
R=mattdr@google.com, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1964663004 .

Cr-Commit-Position: refs/heads/master@{#12681}
2016-05-11 08:23:34 +00:00
Henrik Boström
db7bd3a586 FakeDtlsIdentityStore supporting both RSA and ECDSA.
Previously it only supported RSA-1024/0x10001, now it also supports ECDSA-P256.
This will be necessary for when KT_DEFAULT changes from KT_RSA to KT_ECDSA
since FakeDtlsIdentityStore is used by many tests.

BUG=webrtc:5795
R=mattdr@google.com, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1965723002 .

Cr-Commit-Position: refs/heads/master@{#12680}
2016-05-11 08:20:57 +00:00
pbos
b6e8f2f7a7 Reland of name OpenH264 frame-type conversion function. (patchset #1 id:1 of https://codereview.webrtc.org/1964913002/ )
Reason for revert:
Not perf-regression culprit.

Original issue's description:
> Revert of Rename OpenH264 frame-type conversion function. (patchset #2 id:20001 of https://codereview.webrtc.org/1943193003/ )
>
> Reason for revert:
> Speculative revert for perf regression (though unlikely).
>
> Original issue's description:
> > Rename OpenH264 frame-type conversion function.
> >
> > Also removing default case, so if another frame is added to
> > EVideoFrameType we have to handle it.
> >
> > This will now NOTREACHED on videoFrameTypeInvalid, but
> > videoFrameTypeInvalid shouldn't happen if encoding succeeds, so it
> > should be fine or we should become aware of it.
> >
> > BUG=
> > R=hbos@webrtc.org
> >
> > Committed: https://crrev.com/39a36705ab734914d500b8a0f214ea630d82ab70
> > Cr-Commit-Position: refs/heads/master@{#12636}
>
> TBR=hbos@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:610347
>
> Committed: https://crrev.com/1abf937cecea56ee02ac4a08980ffea9e7ed1054
> Cr-Commit-Position: refs/heads/master@{#12677}

TBR=hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:610347

Review-Url: https://codereview.webrtc.org/1970513004
Cr-Commit-Position: refs/heads/master@{#12679}
2016-05-11 07:58:42 +00:00
nisse
4996eaa7a2 Revert of Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://cod… (patchset #1 id:1 of https://codereview.webrtc.org/1944273002/ )
Reason for revert:
Downstream users updated now.

Original issue's description:
> Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ )
>
> Reason for revert:
> This CL breaks the google3 import (but not the import bot).
> This partial revert only reverts the build files. A full revert no longer cleanly applies to ToT, so this was done instead.
>
> Original issue's description:
> > Enable -Winconsistent-missing-override flag.
> >
> > The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
> >
> > NOPRESUBMIT=True
> > BUG=webrtc:3970
> >
> > Committed: https://crrev.com/ef8b61e11062295365f11b9942f18a08a8b3ec60
> > Cr-Commit-Position: refs/heads/master@{#12563}
>
> TBR=mflodman@webrtc.org,kjellander@webrtc.org,nisse@webrtc.org
> BUG=webrtc:3970
>
> Committed: https://crrev.com/053f91774149a5367ddd531999d4ca69a57dbaa3
> Cr-Commit-Position: refs/heads/master@{#12624}

TBR=kjellander@webrtc.org,mflodman@webrtc.org,ivoc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:3970
NOTRY=True

Review-Url: https://codereview.webrtc.org/1959353002
Cr-Commit-Position: refs/heads/master@{#12678}
2016-05-11 06:28:22 +00:00
Peter Boström
1abf937cec Revert of Rename OpenH264 frame-type conversion function. (patchset #2 id:20001 of https://codereview.webrtc.org/1943193003/ )
Reason for revert:
Speculative revert for perf regression (though unlikely).

Original issue's description:
> Rename OpenH264 frame-type conversion function.
>
> Also removing default case, so if another frame is added to
> EVideoFrameType we have to handle it.
>
> This will now NOTREACHED on videoFrameTypeInvalid, but
> videoFrameTypeInvalid shouldn't happen if encoding succeeds, so it
> should be fine or we should become aware of it.
>
> BUG=
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/39a36705ab734914d500b8a0f214ea630d82ab70
> Cr-Commit-Position: refs/heads/master@{#12636}

TBR=hbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:610347

Review URL: https://codereview.webrtc.org/1964913002 .

Cr-Commit-Position: refs/heads/master@{#12677}
2016-05-10 18:52:13 +00:00
minyue-webrtc
79553cb66e Using ring buffer for AudioVector in NetEq.
AudioVector used NetEq was based on a shift buffer, which has a high complexity, and the complexity is very much dependent on the capacity of the buffer.

This CL changes the shift buffer to a ring buffer.

Reduction in the CPU usages of NetEq is expected.

BUG=608644
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1948483002 .

Cr-Commit-Position: refs/heads/master@{#12676}
2016-05-10 17:56:10 +00:00
Sergey Ulanov
17fa67214c Fix AllocationSequence to handle the case when TurnPort stops using shared socket.
AllocationSequence is responsible for receiving incoming packets on
a shared UDP socket and passing them to the Port objects. TurnPort
may stop sharing UDP socket in which case it allocates a new socket.
AllocationSequence::OnReadPacket() wasn't handling that case properly
which was causing an assert in TurnPort::OnReadPacket().

BUG=webrtc:5757
R=honghaiz@webrtc.org, jiayl@chromium.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1871693004 .

Cr-Commit-Position: refs/heads/master@{#12675}
2016-05-10 17:20:54 +00:00
Niels Möller
d28db7fd65 Delete all use of tick_util.h.
Depends on Chrome cl https://codereview.chromium.org/1888003002/, which was landed some time ago.

BUG=webrtc:5740
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1888593004 .

Cr-Commit-Position: refs/heads/master@{#12674}
2016-05-10 14:31:58 +00:00
minyuel
b031a2e862 Allow WebRTC to offer receiving capability for 120ms Opus packets.
TEST=Build Chromium for receiving + a special AppRTCDemo built with 120ms Opus sending capability. Call went well.

BUG=
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1957963002 .

Cr-Commit-Position: refs/heads/master@{#12673}
2016-05-10 13:35:30 +00:00
henrik.lundin
f3995f71ce NetEq: Implement Expand::Muted
Adding a new method to the Expand class, which will answer the question
whether an ongoing expansion has been faded down to zero
amplitude (i.e., been muted). Also adding a test.

This new functionality will be used in CLs to follow.

BUG=webrtc:5608
NOTRY=True

Review-Url: https://codereview.webrtc.org/1967473004
Cr-Commit-Position: refs/heads/master@{#12672}
2016-05-10 12:54:43 +00:00
henrik.lundin
60f6ce2a29 NetEq: Update stats earlier in the GetAudioInternal call
This is to prepare for implementation of NetEq muted state, which may
cause GetAudioInternal to make an early return just before the call to
GetDecision. With this change, the stats are updated in any case.

BUG=webrtc:5608
NOTRY=True

Review-Url: https://codereview.webrtc.org/1948663002
Cr-Commit-Position: refs/heads/master@{#12671}
2016-05-10 10:52:13 +00:00
Henrik Lundin
47b17dc59c NetEq: Replace timescale_holdoff_ with a Countdown timer
The timescale_holdoff_ is a counter in the DecisionLogic class. The
purpose is to enforce a minimum number of GetAudio calls
between (successfull) time-scaling operations (i.e., Accelerate and
Pre-emptive Expand operations). With this change, the counter is
replaced with a Countdown timer obtained from a TickTimer object.

BUG=webrtc:5608
R=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1945863002 .

Cr-Commit-Position: refs/heads/master@{#12670}
2016-05-10 08:21:11 +00:00
danilchap
6eaa3a41ce _boundingSetToSend moved out of tmmbr_help_ into tmmbn_to_send_
because in the TMMBRHelp class it is independent of other members.

BUG=webrtc:5565
R=philipel

Review-Url: https://codereview.webrtc.org/1746773002
Cr-Commit-Position: refs/heads/master@{#12669}
2016-05-09 17:59:55 +00:00
hta
db3eea0ede Fix codec name logging in ivf_file_writer.cc
The logging code was using the wrong constants for the
codec type, resulting in the type always being "unknown".

Tested: modules_unittests --gtest_filter='IvfFile*' -logs

BUG=

Review-Url: https://codereview.webrtc.org/1955273002
Cr-Commit-Position: refs/heads/master@{#12668}
2016-05-09 17:56:37 +00:00
kjellander@webrtc.org
aa551e66a2 Add test annotation to PeerConnectionClientTest.testLoopbackVp9DecodeToTexture test.
Adding the last test, which was missed in
https://codereview.webrtc.org/1962533002/

TBR=phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1961113002 .

Cr-Commit-Position: refs/heads/master@{#12666}
2016-05-09 17:15:24 +00:00
magjed
2aa84260d8 Android: Handle SurfaceTextureHelper ctor failure for decoder and capturer
BUG=webrtc:5874
TEST=Manually throw an exception inside the SurfaceTextureHelper ctor and run AppRTCDemo.

Review-Url: https://codereview.webrtc.org/1840193007
Cr-Commit-Position: refs/heads/master@{#12665}
2016-05-09 15:28:55 +00:00