14320 Commits

Author SHA1 Message Date
perkj
95a226f55a Replace VideoCapturerInput with VideoSinkInterface.
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)

This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.

BUG=webrtc:5687
// Android CQ seems broken.
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257413002
Cr-Commit-Position: refs/heads/master@{#14238}
2016-09-15 15:57:26 +00:00
Danil Chapovalov
91511f13e1 Split RtcpReceiver::HandleSenderReceiverReport into two functions
as a preparation to replace parsing implementation

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

Review URL: https://codereview.webrtc.org/2340763002 .

Cr-Commit-Position: refs/heads/master@{#14237}
2016-09-15 14:24:42 +00:00
nisse
edebf45712 Use I420Buffer rather than VideoFrameBuffer when writing pixels.
Prepares for deleting VideoFrameBuffer::MutableDataY{,U,V}. objc
changes extracted from cl https://codereview.webrtc.org/2278883002/.

BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2346453002
Cr-Commit-Position: refs/heads/master@{#14236}
2016-09-15 14:20:48 +00:00
hbos
8faf9e047e Removed the const char* (StaticString) type from RTCStatsMember.
std::string is all we need. const char* is an annoying special case
because they can't be compared with ==. Having two different string
types was a premature optimization.

BUG=chromium:627816
NOTRY=True

Review-Url: https://codereview.webrtc.org/2340303002
Cr-Commit-Position: refs/heads/master@{#14235}
2016-09-15 13:52:50 +00:00
Magnus Jedvert
4ed5b9f62d Android SurfaceViewRenderer: Create EGL context on render thread
This CL avoids eglMakeCurrent failing on some problematic Marvel based
Jelly Bean devices.

BUG=webrtc:6350
R=perkj@webrtc.org, sakal@webrtc.org

Review URL: https://codereview.webrtc.org/2339573002 .

Cr-Commit-Position: refs/heads/master@{#14234}
2016-09-15 13:30:29 +00:00
maxmorin
ec62374ccd Reland of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2340253003/ )
Reason for revert:
Fix: let audio_device depend on rtc_base on IOS.

Original issue's description:
> Revert of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2346763002/ )
>
> Reason for revert:
> Breaks iOS
>
> Original issue's description:
> > Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base.
> >
> > BUG=webrtc:3806
> > NOTRY=True
> >
> > Committed: https://crrev.com/100c9d02669910bce06099b3cc1eaad60fd661dd
> > Cr-Commit-Position: refs/heads/master@{#14223}
>
> TBR=kjellander@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:3806
>
> Committed: https://crrev.com/89fb9201b70616a1c33e277f38bf9367112536e8
> Cr-Commit-Position: refs/heads/master@{#14224}

TBR=kjellander@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOTRY=true
BUG=webrtc:3806

Review-Url: https://codereview.webrtc.org/2340233003
Cr-Commit-Position: refs/heads/master@{#14233}
2016-09-15 12:11:59 +00:00
stefan
0a6e0dc471 Disable all screen-capturer tests
ScreenCapturerTest.CaptureUpdatedRegion* tests are flaky and are running in the
wrong executable.

BUG=webrtc:6366, chromium:647067
NOTRY=True

Review-Url: https://codereview.webrtc.org/2342823002
Cr-Commit-Position: refs/heads/master@{#14232}
2016-09-15 12:04:41 +00:00
kjellander
17f008bf33 GYP: Remove targets inside include_tests==1 that are converted to GN.
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.

BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
2016-09-15 11:57:39 +00:00
ehmaldonado
8b28b8017f Assume ProjectRootPath() equals ../.. in Desktop
This way we don't have to rely on the existence of DEPS, and the tests
can be run in swarming bots (which don't have a checkout and therefore
don't have a DEPS file).

This seems to be where Chromium is assumming the project root path to
be.

NOTRY=True
BUG=chromium:497757

Review-Url: https://codereview.webrtc.org/2340773002
Cr-Commit-Position: refs/heads/master@{#14230}
2016-09-15 11:45:55 +00:00
philipel
d268d6ffbe Stash non layer-sync frames if we have not yet received an earlier frame for this layer.
In VP8 the assumption is that a non layer-sync frame depends on all previous
frames on temporal layers less or equal to this frames temporal layer.

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

Review URL: https://codereview.webrtc.org/2306513003 .

Cr-Commit-Position: refs/heads/master@{#14229}
2016-09-15 11:43:30 +00:00
kthelgason
ebc34e78db [GN] Add rtc_sdk_framework_objc target to GN
The build artifacts don't look completely identical to the ones generated
by the GYP targets, but manual review shows the same symbols are exported.

On iOS,  the version generated by the GN follows convention, including
a "Headers" directory, and the .modulemap file. I think this is preferred
over the gyp version.

BUG=webrtc:6320
NOTRY=True
TESTED=Run AppRTCDemo on iOS + Mac and verified with nm that they export the same symbols.

Review-Url: https://codereview.webrtc.org/2340633003
Cr-Commit-Position: refs/heads/master@{#14228}
2016-09-15 11:30:21 +00:00
solenberg
11ace15c19 The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it.
The following APIs are removed from VoEAudioProcessing:

  virtual int SetRxNsStatus(int channel,
                            bool enable,
                            NsModes mode = kNsUnchanged) = 0;
  virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode) = 0;
  virtual int SetRxAgcStatus(int channel,
                             bool enable,
                             AgcModes mode = kAgcUnchanged) = 0;
  virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode) = 0;
  virtual int SetRxAgcConfig(int channel, AgcConfig config) = 0;
  virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
  virtual int RegisterRxVadObserver(int channel,
                                    VoERxVadCallback& observer) = 0;
  virtual int DeRegisterRxVadObserver(int channel) = 0;

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2295113002
Cr-Commit-Position: refs/heads/master@{#14227}
2016-09-15 11:29:21 +00:00
kjellander
70d01242f8 MB: Change Android Clang to build shared instead of static.
A recent build error on Android revealed that we're always doing
static builds for the Android Debug builders. This changes one
to be building shared library instead (is_component_build=true),
which should prevent such breakages in the future.

BUG=webrtc:6360
NOTRY=True

Review-Url: https://codereview.webrtc.org/2342033002
Cr-Commit-Position: refs/heads/master@{#14226}
2016-09-15 10:42:34 +00:00
Kári Tristan Helgason
5a20ed36e6 Fix undefined reference to log2 on android
R=nisse@webrtc.org
TBR=sakal@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2341433004 .

Cr-Commit-Position: refs/heads/master@{#14225}
2016-09-15 08:56:30 +00:00
maxmorin
89fb9201b7 Revert of Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base. (patchset #1 id:1 of https://codereview.webrtc.org/2346763002/ )
Reason for revert:
Breaks iOS

Original issue's description:
> Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base.
>
> BUG=webrtc:3806
> NOTRY=True
>
> Committed: https://crrev.com/100c9d02669910bce06099b3cc1eaad60fd661dd
> Cr-Commit-Position: refs/heads/master@{#14223}

TBR=kjellander@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:3806

Review-Url: https://codereview.webrtc.org/2340253003
Cr-Commit-Position: refs/heads/master@{#14224}
2016-09-15 08:45:33 +00:00
maxmorin
100c9d0266 Add arraysize to rtc_base_approved. Remove dependency of audio_device on rtc_base.
BUG=webrtc:3806
NOTRY=True

Review-Url: https://codereview.webrtc.org/2346763002
Cr-Commit-Position: refs/heads/master@{#14223}
2016-09-15 08:40:42 +00:00
kjellander
705ecc5dda GN: Change group deps to public_deps.
During GN vs GYP auditing it was discovered that some
GN targets that had public_configs were not exposing them
to dependents where the dependent depended on a group, which
in turn included that target as a dependency. Instead of
changing those public_configs to all_dependent_configs
(which would be a change from GYP), it's better to just change
those group targets to use public_deps instead.

BUG=webrtc:6323
NOTRY=True
TESTED=Generated GYP and GN project files on Mac and ran the
tools/gyp_flag_compare.py script before and after this patch was
applied. The file in question used for inspection was the
webrtc/api/webrtcsessiondescriptionfactory.cc
which is a part of the libjingle_peerconnection target.

Review-Url: https://codereview.webrtc.org/2344623002
Cr-Commit-Position: refs/heads/master@{#14222}
2016-09-15 07:53:34 +00:00
henrik.lundin
c26f77f5a6 Remove a couple of unnecessary dependencies on gflags
BUG=webrtc:5447, chromium:645069
NOTRY=True

Review-Url: https://codereview.webrtc.org/2335683002
Cr-Commit-Position: refs/heads/master@{#14221}
2016-09-15 07:05:00 +00:00
kjellander
f807a521fa iSAC: Remove unnecessary WEBRTC_LINUX define
Similar to https://codereview.webrtc.org/1539883002
but for GN. This was discovered during GN vs GYP auditing.

NOTRY=True
BUG=webrtc:6323

Review-Url: https://codereview.webrtc.org/2344633002
Cr-Commit-Position: refs/heads/master@{#14220}
2016-09-15 06:15:55 +00:00
nisse
0d14c6abba Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource.
BUG=chromium:645907

Review-Url: https://codereview.webrtc.org/2334683002
Cr-Commit-Position: refs/heads/master@{#14219}
2016-09-14 19:03:21 +00:00
zijiehe
8295c9326d [WebRTC] Add TwoCapturers test and TwoMagnifierCapturers test
This change is to add more test cases for ScreenCapturer implementation.

BUG=

Review-Url: https://codereview.webrtc.org/2320763003
Cr-Commit-Position: refs/heads/master@{#14218}
2016-09-14 17:22:01 +00:00
Danil Chapovalov
3626d7e247 Move CopyOnWriteBuffer functions definitions from .h to .cc
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/2338843002 .

Cr-Commit-Position: refs/heads/master@{#14217}
2016-09-14 15:14:37 +00:00
minyue
2e164c6b53 Adding ChannelController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2319873002
Cr-Commit-Position: refs/heads/master@{#14216}
2016-09-14 13:47:39 +00:00
hbos
fdafab84bc Fix issues with rtc_stats_unittests tests so that they can run on bots.
This target is not run on bots so a couple of issues went under the
radar. If we expose the tests and run them on the bots[1] two issues are
surfaced which this CL fixes. After this CL lands we can enable this
target on the bots without it going red.

rtcstats_unittest.cc: Fix const char* string comparison issue by
comparing with strcmp instead of equality check.

rtcstatscollector_unittest.cc: Fix TSAN issue by constructing
ScopedFakeClock before spawning Threads.

[1] https://codereview.webrtc.org/2340443002/

BUG=chromium:627816
NOTRY=True

Review-Url: https://codereview.webrtc.org/2333343002
Cr-Commit-Position: refs/heads/master@{#14215}
2016-09-14 13:02:20 +00:00
solenberg
6fa69c91d6 Relaxed unnecessarily stringent thread checking in WebRtcAudioSendStream::OnData().
BUG=webrtc:6345

Review-Url: https://codereview.webrtc.org/2332213006
Cr-Commit-Position: refs/heads/master@{#14214}
2016-09-14 13:01:37 +00:00
nisse
cbae0b475c Use I420Buffer rather than VideoFrameBuffer when writing pixels.
Prepares for deleting VideoFrameBuffer::MutableDataY{,U,V}. Android
changes extracted from cl https://codereview.webrtc.org/2278883002/.

BUG=webrtc:5921

Review-Url: https://codereview.webrtc.org/2331383005
Cr-Commit-Position: refs/heads/master@{#14213}
2016-09-14 12:45:31 +00:00
sakal
bc18fc07be Change onCameraOpening to take camera name as a parameter instead of camera id.
Camera id doesn't really exist for Camera2. Changing onCameraOpening to
take a string instead removes ugly code.

BUG=webrtc:6325
R=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2331013002
Cr-Commit-Position: refs/heads/master@{#14212}
2016-09-14 12:36:26 +00:00
kwiberg
9e2be5f292 webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()
Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}
2016-09-14 12:23:29 +00:00
ehmaldonado
3a7f35b1c4 GN: Declare resources for targets.
Declare resources for GN targets so that they can be isolated

NOTRY=True
BUG=chromium:497757

Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
2016-09-14 12:10:06 +00:00
gaetano.carlucci
52a5703721 Enable BWE logging to command line when rtc_enable_bwe_test_logging is set to true
This patch enables bwe related variable logging to the command line.
This is useful to test congestion control algorithm over real networks.

NOTRY=true

Review-Url: https://codereview.webrtc.org/2296253002
Cr-Commit-Position: refs/heads/master@{#14209}
2016-09-14 12:04:43 +00:00
magjed
b471d1cffb Android EglBase: Include EGL error code in exceptions
This CL appends the EGL error code in exceptions after an EGL function
fails. This information is helpful when debugging.

BUG=webrtc:6350

Review-Url: https://codereview.webrtc.org/2338033002
Cr-Commit-Position: refs/heads/master@{#14208}
2016-09-14 09:40:58 +00:00
kthelgason
194f40a2e7 Refactor QualityScaler and MovingAverage
The MovingAverage class was very specific to the QualityScaler. This
commit generalizes the MovingAverage class to be useful in other
situations as well, and adapts the QualityScaler to use the new
MovingAverage.

BUG=webrtc:6304

Review-Url: https://codereview.webrtc.org/2310853002
Cr-Commit-Position: refs/heads/master@{#14207}
2016-09-14 09:15:02 +00:00
nisse
a075848ebd New method TimestampAligner::TranslateTimestamp
Also enforce a minimum inter-frame interval of 1 ms,
fix a bug in the clipping logic, and improve comments.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/2325563002
Cr-Commit-Position: refs/heads/master@{#14206}
2016-09-14 07:37:03 +00:00
maxmorin
f8a4ecc4a1 Remove dependency of audio_device on metrics_default.
BUG=webrtc:6349
NOTRY=True

Review-Url: https://codereview.webrtc.org/2338813002
Cr-Commit-Position: refs/heads/master@{#14205}
2016-09-14 07:20:26 +00:00
danilchap
17366bc090 Remove handling unused rtcp packets.
App, ExtendedJitterReport and VoipMetric in ExtenedReports are not
used when received (no callbacks, no state change), so removed.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2320703003
Cr-Commit-Position: refs/heads/master@{#14204}
2016-09-14 06:54:55 +00:00
nisse
cdf37a9293 Delete Timing class, timing.h, and update all users.
BUG=webrtc:6324

Review-Url: https://codereview.webrtc.org/2290203002
Cr-Commit-Position: refs/heads/master@{#14203}
2016-09-14 06:41:55 +00:00
peah
d29e3ea4b2 Added build flag around the Intelligibility enhancer performance test code
BUG=chromium:641931

Review-Url: https://codereview.webrtc.org/2294093004
Cr-Commit-Position: refs/heads/master@{#14202}
2016-09-14 04:42:43 +00:00
minyue
caa9cb2cea Adding basic implementation of AudioNetworkAdaptor.
The basic implementation of AudioNetworkAdaptor include the introduction of
  1. Controller
  2. ControllerManager

ControllerManager is to hold all needed controllers. It also orders them according to their significance in dealing with current network condition.

Controller provides an interface MakeDecision, which has to be implemented by specific controllers. AudioNetworkAdaptorImpl calls MakeDecision of the controllers in the order decided by ControllerManager to collect EncoderRuntimeConfig.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2306083002
Cr-Commit-Position: refs/heads/master@{#14201}
2016-09-13 20:34:22 +00:00
danilchap
dd12892ede Reland of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2332673003/ )
Reason for revert:
Fuzzer changed not use functions moved to private.

Original issue's description:
> Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ )
>
> Reason for revert:
> Breaks fuzzer compilation.
>
> Original issue's description:
> > Make rtcp parsing implementation private in RtcpReceiver:
> > Function just for Parse and for Callbacks moved to private section.
> > All handles moved from protected to private section.
> >
> > BUG=webrtc:5260
> > R=sprang@webrtc.org
> >
> > Committed: https://crrev.com/faf708e238c7b43a732fbebf79ac9298b4b95a95
> > Cr-Commit-Position: refs/heads/master@{#14181}
>
> TBR=sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5260
>
> Committed: https://crrev.com/180e4525ca7c9a23602cdf37a8756df7d23e7143
> Cr-Commit-Position: refs/heads/master@{#14182}

TBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2336213002
Cr-Commit-Position: refs/heads/master@{#14200}
2016-09-13 19:23:33 +00:00
kwiberg
d59d3bb117 Replace a DCHECK with static_assert
This requires marking a bunch of compile-time constants "constexpr"
instead of just "const".

Review-Url: https://codereview.webrtc.org/2335483003
Cr-Commit-Position: refs/heads/master@{#14199}
2016-09-13 14:49:41 +00:00
solenberg
ba56b6c7d2 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.

Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.

BUG=
NOPRESUBMIT=true

Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
2016-09-13 14:29:19 +00:00
charujain
bb723e53b4 Fixed video_loopback target.
Moved it inside the rtc_include_tests if clause
so that it build only when tests flag is set to true.

NOTRY=True

BUG=31425205

Review-Url: https://codereview.webrtc.org/2337463002
Cr-Commit-Position: refs/heads/master@{#14197}
2016-09-13 12:52:54 +00:00
Danil Chapovalov
2b2779f3f1 Make CopyOnWriteBuffer keep capacity
for SetData and Clear functions too.

This way result of all functions is same
for shared and non-shared buffer cases

R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/2328553002 .

Cr-Commit-Position: refs/heads/master@{#14196}
2016-09-13 12:15:23 +00:00
Danil Chapovalov
9708884c31 Update rtcp receiver fuzzer to use generic function
IncomingPacket(const uint8_t* packet, size_t size)
instead of implementation specific
IncomingRTCPPacket(PacketInfo* out, Parser* in)
This would allow switch parse implementation

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

Review URL: https://codereview.webrtc.org/2334643002 .

Cr-Commit-Position: refs/heads/master@{#14195}
2016-09-13 10:41:53 +00:00
brandtr
6631e8a21b Minor fixes in FEC and RtpSender{,Video}
- Rename GetNumberOfFecPackets -> NumFecPackets and
  PacketOverhead -> MaxPacketOverhead in ForwardErrorCorrection.
- Rename FECPacketOverhead -> FecPacketOverhead in ProducerFec.
- Move ownership of ForwardErrorCorrection from RTPSenderVideo
  to ProducerFec.
- Make MaxPacketOverhead a member function of ForwardErrorCorrection.
  This will allow for changing it, based on FEC header types, later on.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2275443002
Cr-Commit-Position: refs/heads/master@{#14194}
2016-09-13 10:23:34 +00:00
solenberg
07d9e545ff Revert of Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/ (patchset #7 id:120001 of https://codereview.webrtc.org/2319583005/ )
Reason for revert:
Breaks downstream code

Original issue's description:
> Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
>
> Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
>
> Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
>
> BUG=
> NOPRESUBMIT=true
>
> Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
> Cr-Commit-Position: refs/heads/master@{#14191}

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

Review-Url: https://codereview.webrtc.org/2336123002
Cr-Commit-Position: refs/heads/master@{#14193}
2016-09-13 08:24:10 +00:00
kwiberg
22487b2d0a webrtc/base: Use RTC_DCHECK() instead of assert()
Review-Url: https://codereview.webrtc.org/2325623002
Cr-Commit-Position: refs/heads/master@{#14192}
2016-09-13 08:17:19 +00:00
solenberg
ade2a038a9 Moved webrtc/test/channel_transport/ into webrtc/voice_engine/test/
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.

Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.

BUG=
NOPRESUBMIT=true

Review-Url: https://codereview.webrtc.org/2319583005
Cr-Commit-Position: refs/heads/master@{#14191}
2016-09-13 08:10:54 +00:00
peah
88ac853e14 The current scheme for setting parameters and specifying the
behavior of the audio processing module is quite complex and hard to
implement in a threadsafe and efficient manner. Therefore a new
scheme for setting the parameters in the audio processing module is
introduced in this CL.

The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.

TBR=henrik.lundin@webrtc.org, solenberg@webrtc.org,
BUG=webrtc:5298

Review-Url: https://codereview.webrtc.org/2338493002
Cr-Commit-Position: refs/heads/master@{#14190}
2016-09-12 23:47:32 +00:00
Irfan Sheriff
b2540bb99f Probing: Add support for exponential startup probing
Adds support for exponentially probing the bandwidth at start-up to allow
ramp-up to real capacity of the network.

BUG=webrtc:6332
R=philipel@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2235373004 .

Cr-Commit-Position: refs/heads/master@{#14189}
2016-09-12 19:29:05 +00:00