11758 Commits

Author SHA1 Message Date
mbonadei
6ff8f96baf Tightening visibility of video_toolbox_cc.
This CL aligns the visibility to the standard decided in:
https://codereview.webrtc.org/3001623003.

BUG=webrtc:7743
NOTRY=True

Review-Url: https://codereview.webrtc.org/3010123002
Cr-Commit-Position: refs/heads/master@{#19659}
2017-09-04 12:51:34 +00:00
kwiberg
529662a44c Move array_view.h to webrtc/api/
We use ArrayView in our public API, so its header should be in
webrtc/api/.

BUG=none

Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 12:43:17 +00:00
nisse
2c7b7a633f Delete static method TaskQueue::IsCurrent.
The static method takes a queue name as argument. Used only in one
place, a DCHECK in IncomingVideoStream, which is converted to use
the non-static version of IsCurrent.

BUG=webrtc:8166

Review-Url: https://codereview.webrtc.org/3004873002
Cr-Commit-Position: refs/heads/master@{#19657}
2017-09-04 12:18:21 +00:00
mbonadei
334f9e6a8d Tracking mock_paced_sender.h with a GN target
Untracked headers fly under the 'gn check' radar and in the long term
this can cause problems like unnoticed cyclic dependencies.

This cl creates a synthetic target for this header since no other
targets in webrtc/modules/pacing/BUILD.gn seem to be related to it.

BUG=webrtc:7649
NOTRY=True

Review-Url: https://codereview.webrtc.org/2887593003
Cr-Commit-Position: refs/heads/master@{#19656}
2017-09-04 11:57:11 +00:00
brandtr
83e887cdcb Delete video_quality_measurement.
We don't use it and we don't plan on using it.

BUG=none

Review-Url: https://codereview.webrtc.org/3005993002
Cr-Commit-Position: refs/heads/master@{#19655}
2017-09-04 11:52:31 +00:00
kthelgason
ebd4f7988e Let CreateVideoDecoder take a cricket::VideoCodec.
This makes it possible for decoder factories to actually provide any
video codec, not just the ones WebRTC knows about. It also brings
the decoder factory interface more in line with that of the encoder
factory.

BUG=webrtc:8140

Review-Url: https://codereview.webrtc.org/3007433002
Cr-Commit-Position: refs/heads/master@{#19654}
2017-09-04 11:36:21 +00:00
nisse
3c39c0137a Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
Reason for revert:
A few perf tests broken, including

RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
RampUpTest.UpDownUpTransportSequenceNumberRtx
RampUpTest.UpDownUpTransportSequenceNumberPacketLoss

Original issue's description:
> Use RtxReceiveStream.
>
> This also has the beneficial side-effect that when a media stream
> which is protected by FlexFEC receives an RTX retransmission, the
> retransmitted media packet is passed into the FlexFEC machinery,
> which should improve its ability to recover packets via FEC.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3008773002
> Cr-Commit-Position: refs/heads/master@{#19649}
> Committed: 5c0f6c62ea

TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3010983002
Cr-Commit-Position: refs/heads/master@{#19653}
2017-09-04 11:22:15 +00:00
sakal
07a3bd7c4b Bindings for injectable Java video encoders.
BUG=webrtc:7760

Review-Url: https://codereview.webrtc.org/3003873002
Cr-Commit-Position: refs/heads/master@{#19651}
2017-09-04 10:57:21 +00:00
ilnik
75204c5ccd Change reporting of timing frames conditions in GetStats on receive side
Instead of the longest frame since the last GetStats call, the longest
frame received during last 10 seconds should be returned in GetStats().

Previous way is not a good idea because there are maybe several
consumers of GetStats calls. If not all of them parse timing frame
reports, some of them may be lost.

Also, streamline reporting of TimingFrames via GetStats (remove separate
methods and use VideoReceiveStream::Stats struct instead).

BUG=webrtc:7594

Review-Url: https://codereview.webrtc.org/3008983002
Cr-Commit-Position: refs/heads/master@{#19650}
2017-09-04 10:35:40 +00:00
nisse
5c0f6c62ea Use RtxReceiveStream.
This also has the beneficial side-effect that when a media stream
which is protected by FlexFEC receives an RTX retransmission, the
retransmitted media packet is passed into the FlexFEC machinery,
which should improve its ability to recover packets via FEC.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3008773002
Cr-Commit-Position: refs/heads/master@{#19649}
2017-09-04 10:14:40 +00:00
brandtr
ffbe1cd07e Remove VideoProcessorIntegrationTest::SetTestConfig.
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3004983002
Cr-Commit-Position: refs/heads/master@{#19648}
2017-09-04 10:03:40 +00:00
perkj
1f88531038 Revert of Prepare for injectable SW decoders (patchset #3 id:40001 of https://codereview.webrtc.org/3009973002/ )
Reason for revert:
Tentative revert since it seems to cause problems in Chrome, MAC.

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/42684

Original issue's description:
> Prepare for injectable SW decoders
>
> Pretty much mirrors the work done on the encoding side in CLs:
>
> "Clean up ownership of webrtc::VideoEncoder"
> https://codereview.webrtc.org/3007643002/
>
> "Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder"
> https://codereview.webrtc.org/3007683002/
>
> "WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs"
> https://codereview.webrtc.org/3006713002/
>
> BUG=webrtc:7925
>
> Review-Url: https://codereview.webrtc.org/3009973002
> Cr-Commit-Position: refs/heads/master@{#19641}
> Committed: 084c55a63a

TBR=magjed@webrtc.org,andersc@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3010953002
Cr-Commit-Position: refs/heads/master@{#19647}
2017-09-04 09:43:10 +00:00
minyue-webrtc
bf94fda1e4 Renaming probing_interval to bwe_period globally.
This is a follow up of https://codereview.webrtc.org/2888893002/.

Bug: None
Change-Id: Ia76903858c0a6f2801f14878980e18ae6d3b85e6
Reviewed-on: https://chromium-review.googlesource.com/646020
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li (BackIn2018March) <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19644}
2017-09-02 05:00:01 +00:00
Minyue Li
85a3b6b43a Reland "Supporting 48kHz PCM file."
This was first reviewed in https://codereview.webrtc.org/2790493004/.

It got reverted in https://codereview.webrtc.org/2791453004/ due to
upstreaming error.

Bug: None
TBR: niklas.enbom@webrtc.org
Change-Id: Ia5e9bf86e004258b2aa7822bd489d357fcb8f906
Reviewed-on: https://chromium-review.googlesource.com/645634
Reviewed-by: Minyue Li (BackIn2018March) <minyue@webrtc.org>
Commit-Queue: Minyue Li (BackIn2018March) <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19642}
2017-09-01 19:49:24 +00:00
andersc
084c55a63a Prepare for injectable SW decoders
Pretty much mirrors the work done on the encoding side in CLs:

"Clean up ownership of webrtc::VideoEncoder"
https://codereview.webrtc.org/3007643002/

"Let VideoEncoderSoftwareFallbackWrapper own the wrapped encoder"
https://codereview.webrtc.org/3007683002/

"WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs"
https://codereview.webrtc.org/3006713002/

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3009973002
Cr-Commit-Position: refs/heads/master@{#19641}
2017-09-01 17:38:15 +00:00
magjed
85d18d43ad ObjC: Add null checks to HW encoder compressionOutputCallback
This will help debugging.

BUG=b/65254613

Review-Url: https://codereview.webrtc.org/3012693002
Cr-Commit-Position: refs/heads/master@{#19640}
2017-09-01 13:32:57 +00:00
Stefan Holmer
1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 13:29:30 +00:00
brandtr
12a47f6965 Split up VideoProcessorIntegrationTest files.
Changes made:
* VideoProcessorIntegrationTest definition goes in .h file.
* Member function definitions go into .cc file.
* Tests move to _libvpx.cc and _openh264.cc files.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3008913002
Cr-Commit-Position: refs/heads/master@{#19637}
2017-09-01 07:10:27 +00:00
zhihuang
e683c6871f Completed the functionalities of SrtpTransport.
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't know
if the keys are from SDES or DTLS handshake.

BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.

SrtpFilter is used by BaseChannel as a helper for SDES negotiation.

BUG=webrtc:7013

Review-Url: https://codereview.webrtc.org/2997983002
Cr-Commit-Position: refs/heads/master@{#19636}
2017-08-31 23:00:07 +00:00
philipel
4c14009e83 Avoid calling RtpFrameObject::GetCodecHeader twice in the RtpFrameReferenceFinder.
This is good for a few reasons:
 - We avoid grabing a lock twice.
 - We avoid an unnecessary copy.
 - We avoid a race where a packet could potentially be removed before the
   second time we call GetCodecHeader.

BUG=None

Review-Url: https://codereview.webrtc.org/3009833002
Cr-Commit-Position: refs/heads/master@{#19635}
2017-08-31 15:31:45 +00:00
sakal
85d7650ab6 Make PeerConnectionClient non-singleton.
Ownership of EglBase is moved to PeerConnectionClient.

BUG=webrtc:8135

Review-Url: https://codereview.webrtc.org/3007893002
Cr-Commit-Position: refs/heads/master@{#19634}
2017-08-31 15:03:46 +00:00
nisse
da194e79c4 Delete remnants of RTX support in voice_engine.
Receive logic in voe::Channel attempted to handle RTX if
RTPPayloadRegistry::IsRtx returns true, but the audio code never calls
the config methods (SetRtxSsrc or SetRtxPayloadType) required for that
to ever happen.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3006913002
Cr-Commit-Position: refs/heads/master@{#19633}
2017-08-31 14:26:17 +00:00
henrika
9868042b05 Removes unused APIs from the ADM (part II).
Removes:

int32_t SpeakerVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneVolumeStepSize(uint16_t* stepSize)
int32_t MicrophoneBoostIsAvailable(bool* available)
int32_t SetMicrophoneBoost(bool enable)
int32_t MicrophoneBoost(bool* enabled)
int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0)
int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS)
int32_t CPULoad(uint16_t* load)
int32_t StartRawOutputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawOutputFileRecording()
int32_t StartRawInputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize])
int32_t StopRawInputFileRecording()
int32_t ResetAudioDevice()

BUG=webrtc:7306

Review-Url: https://codereview.webrtc.org/3006803002
Cr-Commit-Position: refs/heads/master@{#19632}
2017-08-31 13:47:32 +00:00
tschumim
137cd1d788 Add a readme for the network tester.
BUG=None

Review-Url: https://codereview.webrtc.org/3011653003
Cr-Commit-Position: refs/heads/master@{#19631}
2017-08-31 13:24:56 +00:00
Alex Loiko
890988c9cb Run the ClangTidy analyser on the AudioProcessing submodule of WebRTC.
This CL contains automatically applied fixes suggested by the
ClangTidy analyzer (http://clang.llvm.org/extra/clang-tidy/). The
following kinds of fixes is present:

* renaming variables when the names in the method signature don't
  match the names in the method definition
  (ClangTidy:readability-inconsistent-declaration-parameter-name)

* ClangTidy:readability-container-size-empty,
  ClangTidy:misc-unused-using-decls,
  ClangTidy:performance-unnecessary-value-param,
  ClangTidy:readability-redundant-control-flow

This is a 'pilot' CL to check if automatic code analyzers can
feasibly be integrated into the WebRTC infrastructuve.

The renamings have been manually expected for consistency with 
surrounding code. In echo_cancellation.cc, I changed several names in
the function implementation to match the function declaration. The
tool suggested changing everything to match the function definitions
instead.

Bug: None
Change-Id: Id3b7ba18c51f15b025f26090c7bdcc642e48d8fd
Reviewed-on: https://chromium-review.googlesource.com/635766
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19630}
2017-08-31 12:53:44 +00:00
Gordana Cmiljanovic
5b0690699d [MIPS]: Fix compiler error for mips32r2 dspr2
Fix error reported by llvm:
"error: unused variable 'kResampleAllpass1' [-Werror,-Wunused-const-variable]"
"error: unused variable 'kResampleAllpass2' [-Werror,-Wunused-const-variable]"

BUG=Unused variable.

Change-Id: I60c64b04e0fa96eb803416fb2148a81fc46a337d
Reviewed-on: https://chromium-review.googlesource.com/575132
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19629}
2017-08-31 12:48:24 +00:00
magjed
6ed63255ca Refactor RTX video codec and payload type assignment
We want to reuse some of this functionality for the new video codec
factories, but not of all it, so this CL refactors out what we want to
reuse to a static function.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3010743002
Cr-Commit-Position: refs/heads/master@{#19628}
2017-08-31 12:37:06 +00:00
minyue-webrtc
5fcd1cc4bb Decouple FEC controller's decision from the decision making for frame length.
Bug: webrtc:8098
Change-Id: If313c8069a16bfb3c2752dcca7fb1cfca24c1caf
Reviewed-on: https://chromium-review.googlesource.com/643299
Reviewed-by: Michael T <tschumim@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19627}
2017-08-31 12:08:56 +00:00
ilnik
2ae19db7d0 Fix alr tests config
BUG=chromium:760900,webrtc:8032

Review-Url: https://codereview.webrtc.org/3007863003
Cr-Commit-Position: refs/heads/master@{#19626}
2017-08-31 11:46:45 +00:00
eladalon
ffe2e14183 Allow PostTask() to take unique_ptr to classes derived of QueuedTask
Problem fixed by this CL: Let DerivedQueuedTask be a custom derivation of QueuedTask. Calling PostTask() with a std::unique_ptr<DerivedQueuedTask> does not work, because overload resolution sees PostTask(const Closure& closure) as a better match. The workaround of explicitly converting to std::unique_ptr<QueuedTask> before calling PostTask() results in less readable code.

Solution: Use std::enable_if to limit the template, thereby making the compiler use the right version of PostTask().

BUG=webrtc:8188

Review-Url: https://codereview.webrtc.org/3006933002
Cr-Commit-Position: refs/heads/master@{#19625}
2017-08-31 11:36:05 +00:00
oprypin
6e09d875fb Replace remaining gflags usages with rtc_base/flags
Continued from https://codereview.webrtc.org/2995363002

BUG=webrtc:7644

Review-Url: https://codereview.webrtc.org/3005483002
Cr-Commit-Position: refs/heads/master@{#19624}
2017-08-31 10:21:39 +00:00
asapersson
f2972ba166 Add some unit tests to TestVp8Impl.
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/3005533003
Cr-Commit-Position: refs/heads/master@{#19623}
2017-08-31 10:01:26 +00:00
sakal
e172d89f73 Change capture time format to nanoseconds in EncodedImage.
The millisecond field is deprecated and will be removed once the
dependencies have been updated.

BUG=webrtc:7760

Review-Url: https://codereview.webrtc.org/3010623002
Cr-Commit-Position: refs/heads/master@{#19622}
2017-08-31 09:37:28 +00:00
Diogo Real
05ea2b39e0 Deprecate IceServer constructors and update dependencies
Bug: webrtc:8176
Change-Id: I2ebc0edf1776c49c202a181d7597099e9242c0e7
Reviewed-on: https://chromium-review.googlesource.com/642710
Commit-Queue: Diogo Real <diogor@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19621}
2017-08-31 07:40:55 +00:00
nisse
125e95eaf6 Delete unused method FilesystemInterface::IsAbsent.
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/3012583002
Cr-Commit-Position: refs/heads/master@{#19620}
2017-08-31 07:22:06 +00:00
Zijie He
3aa4afd630 Flip IsWindowOnScreen behavior when native APIs fail
Per discussion in
https://chromium-review.googlesource.com/c/external/webrtc/+/641814, the
behavior of IsWindowOnScreen() functions when native APIs fail should be
flipped. I.e. window is *not* on screen if OS cannot find it.

Bug: chromium:758554
Change-Id: Ife449a5261fcd89c37595e29a0b1802fcf3c42a5
Reviewed-on: https://chromium-review.googlesource.com/644290
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19617}
2017-08-31 02:43:14 +00:00
Steve Anton
169629aca3 Change WebRtcSession to have a vector of channels
This is the first step towards supporting multiple audio/video
channels in PeerConnection/WebRtcSession. For now, there can only
be 0 or 1 channels in the vector. This adds the framework so that
all the other code that assumes a single audio/video channel can
be transitioned one-by-one to multiple channels.

Bug: webrtc:8183
Change-Id: I6456af32d6e3adf7eb83e281e43253ea973c4eb9
Reviewed-on: https://chromium-review.googlesource.com/644222
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19615}
2017-08-31 01:15:23 +00:00
Zijie He
4fe660785b Use GetWindowDrawableRect() instead of GetCroppedWindowRect() in WindowCapturerWin
GetCroppingWindowRect() is too generic and easy to be misused. For example, in
WindowCapturerWin, it's used to calculate the DesktopRect of the target window,
which is inaccurate. See the screenshot in the bug, it wrongly crops borders on
Windows 8+.

So GetCroppingWindowRect() should be removed eventually, the logic should be
placed in CroppingWindowCapturerWin. But since it's still used in the deprecated
logic in MouseCursorMonitorWin, I would prefer to remove this function after
MouseCursorMonitor improvement has been finished.

But before that, WindowCapturerWin should use GetWindowDrawableRect() instead of
GetCroppedWindowRect() to avoid the wrongly cropping behavior on Windows 8+.

Bug: webrtc:8157
Change-Id: I012b663dced8105623c563dbe55ffcb8d7f17f75
Reviewed-on: https://chromium-review.googlesource.com/642092
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19614}
2017-08-31 00:24:54 +00:00
Zijie He
f7f685c9de Rename IsWindowMinimized to IsWindowOnScreen
As discussed in change
https://chromium-review.googlesource.com/c/external/webrtc/+/634043, the name of
IsWindowMinimized() functions is too confusing. IsWindowOnScreen() is preferred,
since it describes the behavior of the functions more accurately.

This change does not flip the default behavior of these functions to avoid a
behavior change. TODO has been added; and the flipping will happen in a
following change.

Bug: chromium:758554
Change-Id: I009c0fa57142756e5c83f76b2a3561253db1b67f
Reviewed-on: https://chromium-review.googlesource.com/641814
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19613}
2017-08-31 00:20:54 +00:00
Steve Anton
774115c8c6 Change ChannelManager to use unique_ptr
Also clarify the ownership of created channels.

Bug: None
Change-Id: I9cbaec177069d27da2b3b48b93af48f705243b4b
Reviewed-on: https://chromium-review.googlesource.com/643950
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19612}
2017-08-30 22:52:33 +00:00
Steve Anton
24efa72e54 Fix RTCP transport not destroyed when channel creation fails
Bug: None
Change-Id: Ic2f1b7899307eff9b2c98805a5a0eb22ca2e062d
Reviewed-on: https://chromium-review.googlesource.com/642458
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19611}
2017-08-30 22:35:18 +00:00
mbonadei
13b9882fd3 The goal of this CL is to separate Obj-C/Obj-C++ code from targets which have
also C++ code (see https://bugs.chromium.org/p/webrtc/issues/detail?id=7743
for more information).

To achieve this we have created 2 targets (desktop_capture_objc and
desktop_capture_generic) and desktop_capture will act as a proxy between these
targets (this way we can avoid a circular dependency between
desktop_capture_generic and desktop_capture_objc).

BUG=webrtc:7743

Review-Url: https://codereview.webrtc.org/2989053002
Cr-Commit-Position: refs/heads/master@{#19607}
2017-08-30 14:24:43 +00:00
peah
4fed3c0b6f Further utilizing the AEC3 config struct for constants
BUG=webrtc:5298

Review-Url: https://codereview.webrtc.org/3007833002
Cr-Commit-Position: refs/heads/master@{#19605}
2017-08-30 13:58:44 +00:00
brandtr
b57f42676e VideoProcessorIntegrationTest: make it runnable on a task queue.
* First do all frame processing, then do all rate control
  statistics calculations. This means that we only need to
  synchronize once.
* Run the VideoProcessor on a task queue, thus supporting Android
  HW codecs.
* Add some unit tests for the VideoProcessor.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2997283002
Cr-Commit-Position: refs/heads/master@{#19604}
2017-08-30 13:29:51 +00:00
peah
cd8b079afb Increased the allowed API call jitter in AEC3
This CL increases the amount of API call jitter that is allowed in AEC3
without causing resets of AEC3. This increase is now possible, as non-
causal alignments will be detected by the newly imposed delay bound.

BUG=8175

Review-Url: https://codereview.webrtc.org/3012553002
Cr-Commit-Position: refs/heads/master@{#19603}
2017-08-30 13:19:11 +00:00
kthelgason
f8084d485b Fix memory leak in VideoToolbox encoder.
We were leaking a fragmentation header object on each frame.

BUG=webrtc:8132

Review-Url: https://codereview.webrtc.org/3004013002
Cr-Commit-Position: refs/heads/master@{#19602}
2017-08-30 11:47:10 +00:00
mbonadei
16adf03d25 Recently we moved webrtc/base to webrtc/rtc_base, so these
directives in our DEPS files are not needed anymore.

Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.

BUG=webrtc:7634
NOTRY=True

Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
2017-08-30 11:45:58 +00:00
magjed
a35df4260f WebRtcVideoEngine: Encapsulate logic for unifying internal and external video codecs
This CL encapsulates the logic for unifying the internal and external
video encoders into a helper class. The purpose is to prepare for
introducing a new video encoder factory interface that inherently
represents all encoders (i.e. both internal and external). A helper
interface EncoderFactoryAdapter is introduced that both the old
WebRtcVideoEncoderFactory and the new factory interface can implement
and serves as common point to leave the rest of the code unchanged.

BUG=webrtc:7925

Review-Url: https://codereview.webrtc.org/3006713002
Cr-Commit-Position: refs/heads/master@{#19600}
2017-08-30 11:21:30 +00:00
nisse
386449971a Fix setting of recovered flag in RtxReceiveStream.
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3005793002
Cr-Commit-Position: refs/heads/master@{#19599}
2017-08-30 11:16:40 +00:00
ilnik
6d5b4d6fe1 Piggybacking simulcast id and ALR experiment id into video content type extension.
Use it to slice UMA video receive statis.

BUG=8032

Review-Url: https://codereview.webrtc.org/2986893002
Cr-Commit-Position: refs/heads/master@{#19598}
2017-08-30 10:32:14 +00:00