107 Commits

Author SHA1 Message Date
zijiehe
c59bf0415a Remove differ from ScreenCapturer implementations
We can use ScreenCapturerDifferWrapper if needed, otherwise ScreenCapturer does
not need to calculate updated region itself, setting to entire screen is enough.

BUG=633802

Review-Url: https://codereview.webrtc.org/2348803003
Cr-Commit-Position: refs/heads/master@{#14377}
2016-09-24 00:54:40 +00:00
kthelgason
c37e9835a7 Add custom info.plist to modules_unittests
This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it.

BUG=webrtc:6403

Review-Url: https://codereview.webrtc.org/2359863003
Cr-Commit-Position: refs/heads/master@{#14354}
2016-09-22 15:00:57 +00:00
minyue
d0ede4493e Adding FecController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2337103006
Cr-Commit-Position: refs/heads/master@{#14351}
2016-09-22 13:20:59 +00:00
ossu
a70695a3e1 Moved Opus-specific payload splitting into AudioDecoderOpus.
The biggest change to NetEq is the move from a primary flag, to a
Priority with two separate levels: one set by RED splitting and one
set by the codec itself. This allows us to unambigously prioritize
"fallback" packets from these two sources. I've chosen what I believe
is the sensible ordering: packets that the codec prioritizes are
chosen first, regardless of if they are secondary RED packets or
not. So if we were to use Opus w/ FEC in RED, we'd only do Opus FEC
decoding if there was no RED packet that could cover the time slot.

With this change, PayloadSplitter now only deals with RED
packets. Maybe it should be renamed RedPayloadSplitter?

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2342443005
Cr-Commit-Position: refs/heads/master@{#14347}
2016-09-22 09:07:03 +00:00
minyue
4aec1d4437 Relanding of "Adding BitrateController to audio network adaptor."
Adding BitrateController to audio network adaptor was first landed in https://codereview.webrtc.org/2334613002/ but reverted in https://codereview.webrtc.org/2352223002/ due to ODR violation.

This CL tries to use namespace trick to solve the ODR problem.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2353293002
Cr-Commit-Position: refs/heads/master@{#14343}
2016-09-22 06:01:34 +00:00
minyue
e35d329315 Adding FrameLengthController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2335163002
Cr-Commit-Position: refs/heads/master@{#14339}
2016-09-21 23:00:38 +00:00
zijiehe
acc39c44bc Use RgbaColor in DesktopFrameGenerator and add RgbaColorTest
This change uses RgbaColor in DesktopFrameGenerator instead of raw uint32_t to
avoid potential endian issues.

BUG=633802

Review-Url: https://codereview.webrtc.org/2334853002
Cr-Commit-Position: refs/heads/master@{#14337}
2016-09-21 19:23:22 +00:00
ossu
7f40ba4414 Moved legacy_encoded_audio_frame into audio_decoder_interface.
audio_decoder.cc depends on LegacyEncodedAudioFrame and
LegacyEncodedAudioFrame depends on AudioDecoder::EncodedAudioFrame, so
there's no clear way to separate them as of now. This error is also
hodling up builds downstream. I expect we'll revisit these
dependencies as part of the upcoming larger restructuring effort.

NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2359763002
Cr-Commit-Position: refs/heads/master@{#14329}
2016-09-21 12:50:45 +00:00
minyue
33b96b3588 Revert of Adding BitrateController to audio network adaptor. (patchset #7 id:140001 of https://codereview.webrtc.org/2334613002/ )
Reason for revert:
ODR violation

Original issue's description:
> Adding BitrateController to audio network adaptor.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/26b039a137be0a8703766f45b546b29323de714f
> Cr-Commit-Position: refs/heads/master@{#14293}

TBR=michaelt@webrtc.org,henrik.lundin@webrtc.org,krasin@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2352223002
Cr-Commit-Position: refs/heads/master@{#14327}
2016-09-21 11:30:23 +00:00
ossu
0d526d558b Moved codec-specific audio packet splitting into decoders.
There's still some code run specifically for Opus w/ FEC. It will be
addressed in a separate CL.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2326003002
Cr-Commit-Position: refs/heads/master@{#14319}
2016-09-21 08:57:36 +00:00
Rasmus Brandt
78db1582e5 Generalize FEC header formatting.
- Split out reading/writing of FEC headers to classes separate
  from ForwardErrorCorrection. This makes ForwardErrorCorrection
  oblivious to what FEC header scheme is used, and lets it focus on
  encoding/decoding the FEC payloads.
- Add unit tests for FEC header readers/writers.
- Split ForwardErrorCorrection::XorPackets into XorHeaders and
  XorPayloads and reuse these functions for both encoding and
  decoding.
- Rename AttemptRecover -> AttemptRecovery in ForwardErrorCorrection.

BUG=webrtc:5654
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2260803002 .

Cr-Commit-Position: refs/heads/master@{#14316}
2016-09-21 07:19:42 +00:00
minyue
3548357e1b Adding SmoothingFilter to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2339523002
Cr-Commit-Position: refs/heads/master@{#14313}
2016-09-21 06:13:16 +00:00
minyue
26b039a137 Adding BitrateController to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2334613002
Cr-Commit-Position: refs/heads/master@{#14293}
2016-09-19 16:56:43 +00:00
minyue
186cd06512 Adding DTX controller to audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2320093002
Cr-Commit-Position: refs/heads/master@{#14260}
2016-09-16 12:54:47 +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
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
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
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
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
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
Erik Språng
78ce619a0c Extract simulcast rate allocation outside of video encoder.
This is a first step to refactor this code.
I'm deprecating https://codereview.webrtc.org/1913073002 and
implementing this in smaller more isolated steps.

BUG=webrtc:5206
R=asapersson@webrtc.org, kjellander@webrtc.org, noahric@chromium.org

Review URL: https://codereview.webrtc.org/2288223002 .

Cr-Commit-Position: refs/heads/master@{#14186}
2016-09-12 14:04:56 +00:00
minyue
7610f85a2b Adding AudioNetworkAdaptor interfaces.
AudioNetworkAdaptor is supposed to facilitate AudioEncoder to adapt to varying network conditions.

This is the first of a sequence of CLs that are to add one implementation of AudioNetworkAdaptor.

This CL illustrates the interfaces of the AudioNetworkAdaptor.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2308573002
Cr-Commit-Position: refs/heads/master@{#14115}
2016-09-07 20:51:59 +00:00
zijiehe
0f49daccbe Reland of [WebRTC] A real ScreenCapturer test (patchset #1 id:1 of https://codereview.webrtc.org/2310953002/ )
Reason for revert:
Resubmit capturer tests

Original issue's description:
> Revert of [WebRTC] A real ScreenCapturer test (patchset #8 id:240001 of https://codereview.webrtc.org/2268093002/ )
>
> Reason for revert:
> ScreenCapturerTest.CaptureUpdatedRegion fails on Win DrMemory Full.
>
> Original issue's description:
> > [WebRTC] A real ScreenCapturer test
> >
> > We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
> > ScreenDrawer interface is added to the code base to draw various shapes on the
> > screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
> > cases, some other changes are included,
> >
> > 1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
> > ScreenDrawer to finish all the pending draws. This function now only sleeps 50
> > milliseconds on X11 and 100 milliseconds on Windows.
> >
> > 2. A Color structure to help handle a big-endian or little-endian safe color and
> > provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
> > DesktopFrameGenerator (in change 2202443002) can use this class to create colors
> > and compare with or paint to a DesktopFrame.
> >
> > 3. ScreenDrawer now uses Color structure instead of uint32_t.
> >
> > BUG=314516
> >
> > TBR=kjellander@chromium.org
> >
> > Committed: https://crrev.com/9d1c54ace0dc9f68da0152aa1ded2a8dba0a43ae
> > Cr-Commit-Position: refs/heads/master@{#14058}
>
> TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,zijiehe@chromium.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=314516
>
> Committed: https://crrev.com/4c44202dc348613695a4b529bbd7c9bdab6195ec
> Cr-Commit-Position: refs/heads/master@{#14071}

TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=644130

Review-Url: https://codereview.webrtc.org/2313653003
Cr-Commit-Position: refs/heads/master@{#14113}
2016-09-07 18:52:28 +00:00
solenberg
88499ecaca Moving/renaming webrtc/common.h.
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.

- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.

BUG=webrtc:5879

Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
2016-09-07 14:34:45 +00:00
Danil Chapovalov
857a8fb861 Remove dedicated unittest file for remb format
RembStatus moved to RtcpSender unittest where it fits better
Creating remb in Compound/ReducedSize modes already covered by RtcpSender unittests.
Parsing remb already covered by RtcpReceiverTest.ReceivesRemb

R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2312853002 .

Cr-Commit-Position: refs/heads/master@{#14088}
2016-09-06 09:41:52 +00:00
zijiehe
fef8653c5a An early analysis shows in DirectX based capturer, Windows API returns larger
dirty region than the real screen change. A similar behavior may happen on other
platforms with damage notification support. So it's better to have an individual
layer to handle the Differ logic, and remove capturing independent logic out of
each ScreenCapturer* implementation.

So this change does following things,
1. Update differ_block to handle variable height. differ_block_sse2 has been
renamed to differ_vector_sse2.

2. A new ScreenCapturerDifferWrapper implementation to help set
DesktopFrame::updated_region(). It uses an underlying ScreenCapturer to do
the real capture work, and updates the updated region of DesktopFrame returned
from OnCaptureResult function.

3. FakeDesktopCapturer and FakeScreenCapturer to generate controllable
DesktopFrame by using DesktopFrameGenerator and DesktopFramePainter.

4. Test ScreenCapturerDifferWrapper by using FakeScreenCapturer.

After this change, we can eventually remove all Differ logic from
ScreenCapturer* implementations, and fix a potential crash bug in
ScreenCapturerLinux class. It wrongly assumes previous_frame() has a same size
as current_frame(). https://goo.gl/3nSqOC

BUG=633802

TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2202443002
Cr-Commit-Position: refs/heads/master@{#14076}
2016-09-05 22:26:40 +00:00
asapersson
4c44202dc3 Revert of [WebRTC] A real ScreenCapturer test (patchset #8 id:240001 of https://codereview.webrtc.org/2268093002/ )
Reason for revert:
ScreenCapturerTest.CaptureUpdatedRegion fails on Win DrMemory Full.

Original issue's description:
> [WebRTC] A real ScreenCapturer test
>
> We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
> ScreenDrawer interface is added to the code base to draw various shapes on the
> screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
> cases, some other changes are included,
>
> 1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
> ScreenDrawer to finish all the pending draws. This function now only sleeps 50
> milliseconds on X11 and 100 milliseconds on Windows.
>
> 2. A Color structure to help handle a big-endian or little-endian safe color and
> provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
> DesktopFrameGenerator (in change 2202443002) can use this class to create colors
> and compare with or paint to a DesktopFrame.
>
> 3. ScreenDrawer now uses Color structure instead of uint32_t.
>
> BUG=314516
>
> TBR=kjellander@chromium.org
>
> Committed: https://crrev.com/9d1c54ace0dc9f68da0152aa1ded2a8dba0a43ae
> Cr-Commit-Position: refs/heads/master@{#14058}

TBR=sergeyu@chromium.org,jamiewalch@chromium.org,kjellander@chromium.org,zijiehe@chromium.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=314516

Review-Url: https://codereview.webrtc.org/2310953002
Cr-Commit-Position: refs/heads/master@{#14071}
2016-09-05 13:55:26 +00:00
ehmaldonado
e9cc686293 GN Templates: Move common_inherited_config to the template.
Remove common_inherited_config from the targets and add it to the
template instead.

BUG=webrtc:6187
NOTRY=True

Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
2016-09-05 13:10:23 +00:00
ehmaldonado
7a2ce0b738 GN Templates: Move common_config to the template.
Remove common_config from the targets' config and add
it to the template instead.

BUG=webrtc:6187
NOTRY=True

Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
2016-09-05 08:35:48 +00:00
zijiehe
9d1c54ace0 [WebRTC] A real ScreenCapturer test
We do not have a real ScreenCapturer test before. And after CL 2210443002, a new
ScreenDrawer interface is added to the code base to draw various shapes on the
screen. This change is to use ScreenDrawer to test ScreenCapturer. Besides test
cases, some other changes are included,

1. A WaitForPendingPaintings() function in ScreenDrawer, to wait for a
ScreenDrawer to finish all the pending draws. This function now only sleeps 50
milliseconds on X11 and 100 milliseconds on Windows.

2. A Color structure to help handle a big-endian or little-endian safe color and
provide functions to compare with DesktopFrame::data(). Both ScreenDrawer and
DesktopFrameGenerator (in change 2202443002) can use this class to create colors
and compare with or paint to a DesktopFrame.

3. ScreenDrawer now uses Color structure instead of uint32_t.

BUG=314516

TBR=kjellander@chromium.org

Review-Url: https://codereview.webrtc.org/2268093002
Cr-Commit-Position: refs/heads/master@{#14058}
2016-09-03 02:10:42 +00:00
ehmaldonado
38a2132b02 GN: Introduce templates.
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.

These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target

Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.

BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
2016-09-02 11:10:41 +00:00
solenberg
f383c5754f - Remove unused unit test webrtc/modules/audio_processing/agc/agc_unittest.cc
- Remove webrtc/tools/agc/test_utils.cc/.h - only used from the above test.
- Remove webrtc/tools/agc/agc_harness.cc - not used anymore.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2299023004
Cr-Commit-Position: refs/heads/master@{#14039}
2016-09-02 09:39:40 +00:00
peah
c435946ec1 In order to ensure that the same code is run in the tests as is otherwise
run it is important that the same build flags are used in the code being
tested. For the debugging functionality inside APM, that was not the case
and this is corrected in this CL.

This CL is chained to the CL https://codereview.webrtc.org/2300813004/

BUG=webrtc:5298

Review-Url: https://codereview.webrtc.org/2307563002
Cr-Commit-Position: refs/heads/master@{#14031}
2016-09-02 05:03:32 +00:00
kjellander
32c4a20df7 GN: Fix resource files for iOS test target
In order to get resource files to be properly packaged into
the .app for a unit test on iOS, the resource files needs
to be listed as sources in a bundle_data target.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2292853002
Cr-Commit-Position: refs/heads/master@{#13968}
2016-08-30 09:53:54 +00:00
peah
1bcfce5ff2 Deactivated the intelligibility enhancement functionality by default
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2272423003
Cr-Commit-Position: refs/heads/master@{#13937}
2016-08-26 14:16:13 +00:00
kjellander
28a0ffdd52 GN: Synchronize resources between Android and iOS.
iOS tests packaged into an .app uses the same way of
defining resources (the data attribute). Some iOS
simulator tests are failing due to missing resources, so
let's sync them all.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2277753003
Cr-Commit-Position: refs/heads/master@{#13898}
2016-08-24 14:48:48 +00:00
kwiberg
619a211562 iLBC: Handle a case of bad input data
We detect an unreasonable state (caused by a bad encoded stream)
before it can lead to problems, and handle it by resetting the
decoder.

NOPRESUBMIT=true
BUG=chromium:617124

Review-Url: https://codereview.webrtc.org/2255203002
Cr-Commit-Position: refs/heads/master@{#13888}
2016-08-24 09:46:48 +00:00
isheriff
8df4d0e426 Add playout_delay_oracle_unittest as gn target
BUG=

Review-Url: https://codereview.webrtc.org/2256743002
Cr-Commit-Position: refs/heads/master@{#13821}
2016-08-18 14:53:44 +00:00
peah
e9a6acfbf5 Added missing unittest to the modules/BUILD.gn build file
NOTRY=True

BUG=

Review-Url: https://codereview.webrtc.org/2255093002
Cr-Commit-Position: refs/heads/master@{#13813}
2016-08-18 09:41:51 +00:00
zijiehe
49c01d7f34 Currently there is not way to programmically test whether a ScreenCapturer
implementation can accurately capture updated regions. Especially in
ScreenCapturerWinDirectx, which has a specific updated region spreading logic
and cannot be tested through regular code path. So we need a controllable
ScreenDrawer to draw some basic shapes on the screen. And a platform independent
test case can use the ScreenDrawer to test a ScreenCapturer.

So this change addes a ScreenDrawer virtual class, and its Windows
implementation ScreenDrawerWin. A disabled gtest ScreenDrawerTest.DrawRectangles
is also added to manually test whether ScreenDrawer can work on a certain
platform.

BUG=314516

TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2210443002
Cr-Commit-Position: refs/heads/master@{#13788}
2016-08-17 00:34:00 +00:00
kwiberg
9d7eb13c40 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ )
Reason for revert:
Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This has been landed twice before, as
> https://codereview.webrtc.org/2037623002 and
> https://codereview.webrtc.org/2240163002. Third time's a charm!)
>
> NOPRESUBMIT=True
> TBR=kjellander@webrtc.org
>
> Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef
> Cr-Commit-Position: refs/heads/master@{#13777}

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

Review-Url: https://codereview.webrtc.org/2245413002
Cr-Commit-Position: refs/heads/master@{#13779}
2016-08-16 11:08:39 +00:00
kwiberg
427ce3d86f Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

(This has been landed twice before, as
https://codereview.webrtc.org/2037623002 and
https://codereview.webrtc.org/2240163002. Third time's a charm!)

NOPRESUBMIT=True
TBR=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2247033003
Cr-Commit-Position: refs/heads/master@{#13777}
2016-08-16 10:34:50 +00:00
kwiberg
c8c71f484e Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ )
Reason for revert:
Breaks downstream code, so revert again. Yay.

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> (This is a re-land of https://codereview.webrtc.org/2037623002, which
> had to be reverted.)
>
> NOPRESUBMIT=True
>
> Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1
> Cr-Commit-Position: refs/heads/master@{#13757}

TBR=perkj@webrtc.org,kjellander@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/2245153002
Cr-Commit-Position: refs/heads/master@{#13758}
2016-08-15 18:43:56 +00:00
kwiberg
dc65ea29b3 Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

(This is a re-land of https://codereview.webrtc.org/2037623002, which
had to be reverted.)

NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2240163002
Cr-Commit-Position: refs/heads/master@{#13757}
2016-08-15 17:36:38 +00:00
sakal
714dd4e532 GN: Update tests to have the correct shard timeout value on Android.
TBR=mflodman@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2231413002
Cr-Commit-Position: refs/heads/master@{#13741}
2016-08-15 09:29:19 +00:00
philipel
17deeb47ed PacketBuffer is now ref counted.
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.

BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2199133004 .

Cr-Commit-Position: refs/heads/master@{#13725}
2016-08-11 13:09:40 +00:00
sakal
bd59c71ff8 GN: Add dependency libjingle_peerconnection_java to modules_unittests.
This change only affects the Android build.

BUG=webrtc:6189
TBR=tommi@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2233903002
Cr-Commit-Position: refs/heads/master@{#13722}
2016-08-11 07:59:23 +00:00
philipel
bf8a2c94ce Probe bitrate estimator correction.
Since the interval between the timestamps does not include the send/receive
time of the last/first packet we correct the interval by adding the average
of the interval between probing packets.

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

Review URL: https://codereview.webrtc.org/2224173003 .

Cr-Commit-Position: refs/heads/master@{#13719}
2016-08-10 17:00:52 +00:00
ehmaldonado
f98dc105ba GN: Add target for modules_tests.
Additional changes I needed to make it work:

- Modified a header in RTPFile.cc. Every other file is
  using "webrtc/engine_configurations.h" instead.

- Disabled flag 4373 for msvs because it was disabled
  in build/common.gypi.

BUG=webrtc:6038
TBR=kwiberg@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2187563005
Cr-Commit-Position: refs/heads/master@{#13628}
2016-08-03 17:46:51 +00:00
aleloi
09f45108c2 Removed callback in old AudioConferenceMixer.
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.

Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
2016-07-28 10:52:23 +00:00