3747 Commits

Author SHA1 Message Date
brucedawson
07df20b51a Fix building of bwe_test_logging.cc in gyp and gn builds
bwe_test_logging.cc is supposed to be conditionally built in gyp builds
but, due to a path error in the sources! expressions it was always
compiled.

Meanwhile, compilation of bwe_test_logging.cc was never set up for gn
builds.

This fixes both of these problems.

BUG=604060

Review-Url: https://codereview.webrtc.org/1990373002
Cr-Commit-Position: refs/heads/master@{#12842}
2016-05-23 00:15:11 +00:00
danilchap
6a1a8ce668 [rtcp] in Tmmbn/tmmbr packets bitrate updated to 64bit
to match packet format.

Review-Url: https://codereview.webrtc.org/1991223002
Cr-Commit-Position: refs/heads/master@{#12837}
2016-05-21 16:54:50 +00:00
zijiehe
2970c2a237 DirectX based screen capturer logic
BUG=595530,314516

Review-Url: https://codereview.webrtc.org/1845113002
Cr-Commit-Position: refs/heads/master@{#12835}
2016-05-21 05:08:08 +00:00
minyue
cac94aae3e Clang format on AGC legacy code.
BUG=

Review-Url: https://codereview.webrtc.org/1998183002
Cr-Commit-Position: refs/heads/master@{#12831}
2016-05-20 15:42:30 +00:00
asapersson
01d70a3978 Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h.
Updated tests to use the default implementation and removed the test implementation (webrtc/test/histograms.h).

BUG=

Review-Url: https://codereview.webrtc.org/1915523002
Cr-Commit-Position: refs/heads/master@{#12829}
2016-05-20 13:29:50 +00:00
kwiberg
2036135620 AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo
When this class was created, we inadvertently set the default to 64
kbit/s for both cases, failing to preserve the previous behavior. This
patch restores the old behavior.

From what I've been able to dig up, this problem did not affect Opus
encoders created internally in the Audio Coding Module. Those have
always used the bitrate from the supplied CodecInst.

Review-Url: https://codereview.webrtc.org/1942193002
Cr-Commit-Position: refs/heads/master@{#12827}
2016-05-20 11:57:08 +00:00
kwiberg
3fc8e8dde3 Fix UBSan errors (left shift of negative value)
BUG=chromium:604803

Review-Url: https://codereview.webrtc.org/1988113003
Cr-Commit-Position: refs/heads/master@{#12826}
2016-05-20 11:40:33 +00:00
philipel
94616b3b0b Disabled flaky test TestFrameBuffer2.OneLayerStreamReordered
TBR=stefan@webrtc.org

BUG=

Review URL: https://codereview.webrtc.org/2000573002 .

Cr-Commit-Position: refs/heads/master@{#12821}
2016-05-20 08:43:50 +00:00
minyue
30629957e1 Fix UBSan errors (left shift on negative value)
BUG=webrtc:5530

Review-Url: https://codereview.webrtc.org/1999653002
Cr-Commit-Position: refs/heads/master@{#12820}
2016-05-20 08:23:55 +00:00
ossu
e725f7c73e Turned AudioDecoderFactory into a RefCounted thing to use with scoped_refptr.
First step of AudioDecoderFactory injection CLs. AudioDecoderFactories will be shared, and shared_ptr is currently off the table, so this CL changes the current uses of AudioDecoderFactory from std::unique_ptr to rtc::scoped_refptr.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1990803004
Cr-Commit-Position: refs/heads/master@{#12815}
2016-05-19 17:48:12 +00:00
philipel
6e8224f4ab Reduce flakyness of timing dependent tests for TestFrameBuffer2.*.
BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1994623003 .

Cr-Commit-Position: refs/heads/master@{#12814}
2016-05-19 15:07:56 +00:00
philipel
be7a9e5f8a Revert "Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )"
Also disabled modules_unittest.TestFrameBuffer2.* in drmemory.

This reverts commit b711f10d9683b9de6ee78186f77b225fc7ebfb8f.

TBR=honghaiz@webrtc.org

BUG=

Review URL: https://codereview.webrtc.org/1991133003 .

Cr-Commit-Position: refs/heads/master@{#12806}
2016-05-19 10:19:44 +00:00
honghaiz
b711f10d96 Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )
Reason for revert:
Two tests added by this CL failed in Win DrMemory Full:
 TestFrameBuffer2.OneLayerStreamReordered - TestFrameBuffer2.WaitForFrame

See the link here:
https://build.chromium.org/p/client.webrtc/waterfall?builder=Win%20DrMemory%20Full

Original issue's description:
> FrameBuffer for the new jitter buffer.
>
> BUG=webrtc:5514
> R=danilchap@webrtc.org, mflodman@webrtc.org
>
> Committed: https://crrev.com/a376e70cf9d0df3c35d53533b454da542661775b
> Cr-Commit-Position: refs/heads/master@{#12798}

TBR=mflodman@webrtc.org,danilchap@webrtc.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/1991513004
Cr-Commit-Position: refs/heads/master@{#12800}
2016-05-18 22:52:36 +00:00
philipel
a376e70cf9 FrameBuffer for the new jitter buffer.
BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1969403007 .

Cr-Commit-Position: refs/heads/master@{#12798}
2016-05-18 16:10:14 +00:00
Stefan Holmer
a3c2c3e606 Remove unneccessary log in PacedSender.
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1990883002 .

Cr-Commit-Position: refs/heads/master@{#12796}
2016-05-18 14:06:02 +00:00
henrik.lundin
83e8286c6b AEC: Add UMA logging of buffer re-alignment
This change adds a UMA log that will be written to when a non-zero delay
correction is done in the AEC. The number of elements moved (positive or
negative) will be logged to
"WebRTC.Audio.AecDelayAdjustmentAgnosticValue" or
"WebRTC.Audio.AecDelayAdjustmentSystemValue", depending on whether
delay-agnostic AEC is used or not, respectively.

BUG=webrtc:5903

Review-Url: https://codereview.webrtc.org/1991723002
Cr-Commit-Position: refs/heads/master@{#12795}
2016-05-18 12:43:05 +00:00
henrik.lundin
42dda50860 Propagate muted info from VoE Channel to AudioConferenceMixer
Required updating of a few related classes and tests.

BUG=webrtc:5609
NOTRY=True

Review-Url: https://codereview.webrtc.org/1986093002
Cr-Commit-Position: refs/heads/master@{#12794}
2016-05-18 12:36:07 +00:00
henrik.lundin
5a216d0489 Add muted parameter to audio_frame_manipulator methods
BUG=webrtc:5609
NOTRY=True

Review-Url: https://codereview.webrtc.org/1983023002
Cr-Commit-Position: refs/heads/master@{#12788}
2016-05-18 09:34:24 +00:00
kwiberg
89f237cedc Fix UBSan errors (left shift of negative value, left shift overflows int)
BUG=chromium:603498

Review-Url: https://codereview.webrtc.org/1979973003
Cr-Commit-Position: refs/heads/master@{#12787}
2016-05-18 09:20:35 +00:00
henrik.lundin
60200d1094 Add FrameAndMuteInfo to AudioConferenceMixer
BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1980343002
Cr-Commit-Position: refs/heads/master@{#12786}
2016-05-18 08:48:46 +00:00
Sergey Ulanov
837dde9f1d Remove DesktopFrame::shape().
Frame shape is no longer used and can be removed.

R=jamiewalch@chromium.org

Review URL: https://codereview.webrtc.org/1986963002 .

Cr-Commit-Position: refs/heads/master@{#12783}
2016-05-18 05:03:47 +00:00
henrik.lundin
3a0a0f4b0d AudioConfMixer: Add muted variable to ParticipantFramePair
Renaming the ParticipantFramePair to ParticipantFrameStruct. The muted
variable is not yet used.

BUG=webrtc:5609

Review-Url: https://codereview.webrtc.org/1981243002
Cr-Commit-Position: refs/heads/master@{#12780}
2016-05-18 00:16:05 +00:00
henrik.lundin
d4ccb00b9e Propagate muted parameter to VoE::Channel
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.

BUG=webrtc:5609
TBR=perkj@webrtc.org

Review-Url: https://codereview.webrtc.org/1985743002
Cr-Commit-Position: refs/heads/master@{#12779}
2016-05-17 19:22:03 +00:00
pasko
e305d956c0 Remove runtime NEON detection
Chrome does not detect NEON instruction set at runtime in WebRTC code starting
with M50, which is now in Stable. Remove support for runtime detection for
simplicity.

The only remaining piece of Chrome that will continue to depend on runtime
detection is /net, where devices with _broken_ neon support are also detected,
and it is not configurable via GYP/GN.

BUG=522035
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/1955413003
Cr-Commit-Position: refs/heads/master@{#12778}
2016-05-17 17:56:48 +00:00
kwiberg
9b2228fdfc Fix UBSan errors (left shift of negative value)
BUG=chromium:603501

Review-Url: https://codereview.webrtc.org/1988723002
Cr-Commit-Position: refs/heads/master@{#12775}
2016-05-17 13:40:48 +00:00
nisse
c9c142f170 Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
Reason for revert:
Should work after cl https://codereview.webrtc.org/1985693002/ is landed, which initializes the frames used by FakeWebRtcVideoCaptureModule. So intend to reland after that, with no changes.

Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
>
> Reason for revert:
> Speculative revert to see if failures on the DrMemory bot are related to this cl.  See e.g. here:
> https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
>
> UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
> # 0 CopyRow_AVX
> # 1 CopyPlane
> # 2 I420Copy
> # 3 webrtc::ExtractBuffer
> # 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
> # 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
> # 6 FakeWebRtcVideoCaptureModule::SendFrame
> # 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
> # 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
>
> Original issue's description:
> > Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
> >
> > Reason for revert:
> > I plan to reland this change in a week or two, after downstream users are updated.
> >
> > Original issue's description:
> > > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> > >
> > > Reason for revert:
> > > Breaks chrome FYI bots.
> > >
> > > Original issue's description:
> > > > Delete webrtc::VideoFrame methods buffer and stride.
> > > >
> > > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > > to not imply an AddRef.
> > > >
> > > > BUG=webrtc:5682
> > >
> > > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5682
> > >
> > > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > > Cr-Commit-Position: refs/heads/master@{#12558}
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> > Cr-Commit-Position: refs/heads/master@{#12721}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d49c30cd2fe442f2b5b4ecec8d5cbaa430464725
> Cr-Commit-Position: refs/heads/master@{#12745}

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1979193003
Cr-Commit-Position: refs/heads/master@{#12773}
2016-05-17 11:05:51 +00:00
Alejandro Luebs
c9b0c26e0c Surface the IntelligibilityEnhancer on MediaConstraints
R=henrika@webrtc.org, peah@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1952123003 .

Cr-Commit-Position: refs/heads/master@{#12763}
2016-05-16 22:32:45 +00:00
aleloi
8bce67b745 Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created.
Review-Url: https://codereview.webrtc.org/1967503002
Cr-Commit-Position: refs/heads/master@{#12760}
2016-05-16 14:34:31 +00:00
kwiberg
fd8be3468a Remove webrtc/base/scoped_ptr.h
This is a re-land of https://codereview.webrtc.org/1942823002

TBR=tommi@webrtc.org
BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1966423002
Cr-Commit-Position: refs/heads/master@{#12750}
2016-05-15 02:44:18 +00:00
tommi
d49c30cd2f Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #2 id:290001 of https://codereview.webrtc.org/1963413004/ )
Reason for revert:
Speculative revert to see if failures on the DrMemory bot are related to this cl.  See e.g. here:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243

UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
# 0 CopyRow_AVX
# 1 CopyPlane
# 2 I420Copy
# 3 webrtc::ExtractBuffer
# 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
# 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
# 6 FakeWebRtcVideoCaptureModule::SendFrame
# 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
# 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>

Original issue's description:
> Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
>
> Reason for revert:
> I plan to reland this change in a week or two, after downstream users are updated.
>
> Original issue's description:
> > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> >
> > Reason for revert:
> > Breaks chrome FYI bots.
> >
> > Original issue's description:
> > > Delete webrtc::VideoFrame methods buffer and stride.
> > >
> > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > to not imply an AddRef.
> > >
> > > BUG=webrtc:5682
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > Cr-Commit-Position: refs/heads/master@{#12558}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> Cr-Commit-Position: refs/heads/master@{#12721}

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1983583002
Cr-Commit-Position: refs/heads/master@{#12745}
2016-05-14 10:18:13 +00:00
perkj
fea93099f0 This reland https://codereview.webrtc.org/1932683002/.
Remove ViEEncoder::SetNetworkStatus.

Original cl description:
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.

Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
2016-05-14 07:58:54 +00:00
Peter Boström
1299615838 Make sure WebRTC works without libvpx VP9 support.
Wires up existing libvpx_build_vp9==0 GYP flag into WebRTC and makes VP9
optional. Change is GYP only for now since libvpx's GN files build VP9
unconditionally.

BUG=webrtc:5884
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1970343002 .

Cr-Commit-Position: refs/heads/master@{#12741}
2016-05-14 00:03:28 +00:00
zijiehe
99f8cd0ae2 CURSORINFO.flags should be checked before capturing its bitmap
BUG=566178, 428886

Review-Url: https://codereview.webrtc.org/1959863002
Cr-Commit-Position: refs/heads/master@{#12739}
2016-05-13 23:41:59 +00:00
pbos
1f53452ca6 Unify hardware and software QP thresholds.
Uses current libvpx (slightly older) thresholds to maintain a larger
window of stable QP, but maintains the newer H264 thresholds.

BUG=
R=glaznev@webrtc.org, mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1966213002
Cr-Commit-Position: refs/heads/master@{#12734}
2016-05-13 18:05:38 +00:00
philipel
02447bc408 Logic for finding frame references moved from PacketBuffer to new class
RtpFrameReferenceFinder.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/1961053002
Cr-Commit-Position: refs/heads/master@{#12725}
2016-05-13 13:01:11 +00:00
nisse
d0dc66e0ea Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
Reason for revert:
I plan to reland this change in a week or two, after downstream users are updated.

Original issue's description:
> Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
>
> Reason for revert:
> Breaks chrome FYI bots.
>
> Original issue's description:
> > Delete webrtc::VideoFrame methods buffer and stride.
> >
> > To make the HasOneRef/IsMutable hack work, also had to change the
> > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > to not imply an AddRef.
> >
> > BUG=webrtc:5682
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> Cr-Commit-Position: refs/heads/master@{#12558}

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1963413004
Cr-Commit-Position: refs/heads/master@{#12721}
2016-05-13 11:12:48 +00:00
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
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
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
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
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