367 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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
kjellander
d8dd190a08 GN: Fix test_support_unittests and MIPS compile issue.
Move the webrtc/test/test_support/metrics sources into
test_support[_unittests] targets.
This is essentially reverting https://webrtc-codereview.appspot.com/5789004
and moving these sources back to the right target.

Add missing foreman_cif.yuv resource needed for these tests.

For MIPS, a compile error was surfacing for logcat_trace_context.h when
flipping bot to GN, which was fixed.

BUG=webrtc:5949
NOTRY=True

Review-Url: https://codereview.webrtc.org/2267113002
Cr-Commit-Position: refs/heads/master@{#13860}
2016-08-23 11:52:19 +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
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
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
philipel
233c4ba4fd New ProbingCalculator class.
The ProbingCalculator class calculates and validates the results from
probing attempts.

BUG=webrtc:5859

Review-Url: https://codereview.webrtc.org/2121183002
Cr-Commit-Position: refs/heads/master@{#13589}
2016-08-01 15:49:13 +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
stefan
5e12d36ba7 Reset InterArrival if arrival time clock makes a jump.
Also adds a copy of the BWE test suite to the new DelayBasedBwe class.

BUG=webrtc:6079

Review-Url: https://codereview.webrtc.org/2126793002
Cr-Commit-Position: refs/heads/master@{#13428}
2016-07-11 08:44:12 +00:00
aleloi
77ad394fa6 A simple copy of the old audio mixer to a new directory.
I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer.

This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2104363003
Cr-Commit-Position: refs/heads/master@{#13378}
2016-07-04 13:33:09 +00:00
mflodman
e15032c750 Remove all old suspension logic.
I'm also removing media_optimization_unittest.cc, since it only tested the
suspension logic and nothing else.

R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/2119503002 .

Cr-Commit-Position: refs/heads/master@{#13355}
2016-07-01 07:00:19 +00:00
Alejandro Luebs
a181c9ad17 Keep track of the user-facing number of channels in a ChannelBuffer
Before this change the ChannelBuffer had a fixed number of channels. This meant for example that when the Beamformer would reduce the number of channels to one, the merging filter bank was still merging all the channels, which was unnecessary since they were not processed and just discarded later. This change doesn't change the signal at all. It just reflects the number of channels in the ChannelBuffer, reducing the complexity.

R=henrik.lundin@webrtc.org, peah@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2053773002 .

Cr-Commit-Position: refs/heads/master@{#13352}
2016-06-30 22:33:47 +00:00
peah
ca4cac7e74 New module for the adaptive level controlling functionality in the audio processing module
NOTRY=true
TBR=aluebs@webrtc.org
BUG=webrtc:5920

Review-Url: https://codereview.webrtc.org/2090583002
Cr-Commit-Position: refs/heads/master@{#13333}
2016-06-29 22:26:19 +00:00
kwiberg
e7edea9759 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@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/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
2016-06-22 23:29:58 +00:00
Karl Wiberg
65874b163e Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

R=perkj@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/2037623002 .

Cr-Commit-Position: refs/heads/master@{#13261}
2016-06-22 21:47:53 +00:00
philipel
863a8264cc Use |probe_cluster_id| to cluster packets.
Introduced new class DelayBasedProbingEstimator which is a copy of
RemoteBitrateEstimatorAbsSendTime with only minor changes. This makes probing
more reliable but is still not usable for mid-call probing.

BUG=

Review-Url: https://codereview.webrtc.org/2038023002
Cr-Commit-Position: refs/heads/master@{#13195}
2016-06-17 16:21:43 +00:00
Niels Möller
fc3a8ee47b Delete unused code.
* Unused audio_coding and video_coding test code.
* Obsolete voice_engine android test app.
* Left-over placeholder files for remoteaudiotrack and
  portallocatorfactory.

In addition, change modules.gyp dependency from rtc_base to
rtc_base_approved.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/2065353002 .

Cr-Commit-Position: refs/heads/master@{#13166}
2016-06-16 13:51:40 +00:00
kjellander
fb11424551 GN: Add modules_unittests
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
  * webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
  * webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}

BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.

NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
2016-06-13 07:19:53 +00:00
kjellander
5c1d043726 Fix GYP/GN for webrtc/modules/remote_bitrate_estimator
Sync the GYP and GN targets and update the name of the GN one
to 'remote_bitrate_estimator'.
Move the GYP variable 'enable_bwe_test_logging' into the local scope.
Remove redundant entries in modules.gyp.

These are preparations related to the GN migration.

BUG=webrtc:5949
TESTED=Ran GYP with the default variables and with
-Denable_bwe_test_logging=1. Compiled remote_bitrate_estimator
and verified that bwe_test_logging.cc is compiled only when
set.
NOTRY=True

Review-Url: https://codereview.webrtc.org/2040313004
Cr-Commit-Position: refs/heads/master@{#13087}
2016-06-09 09:41:02 +00:00
philipel
bde418d84c Renamed video_coding/packet_buffer_unittest.cc.
Renamed video_coding/packet_buffer_unittest.cc to
video_coding/video_packet_buffer_unittest.cc

BUG=webrtc:5949
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2049693002
Cr-Commit-Position: refs/heads/master@{#13074}
2016-06-08 19:09:45 +00:00
henrik.lundin
919518613f NetEq: Rename Nack to NackTracker to avoid name collisions in GN
BUG=webrtc:5949
NOTRY=True
R=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2045243002
Cr-Commit-Position: refs/heads/master@{#13069}
2016-06-08 13:43:49 +00:00
peah
bbe423312d Change name of files and class in agc/histogram* in order to avoid issue file-name clash in build files
The changes are done in several patches in order to make
the review easier.

NOTRY=True
BUG=webrtc:5949

Review-Url: https://codereview.webrtc.org/2051443002
Cr-Commit-Position: refs/heads/master@{#13068}
2016-06-08 13:42:08 +00:00
isheriff
6b4b5f3770 Add sender controlled playout delay limits
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.

The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.

There are no tests at this time and most of testing is done with chromium
webrtc prototype.

On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.

BUG=webrtc:5895

Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
2016-06-08 07:24:30 +00:00
Per
69b332df83 Move logic for calculating needed bitrate overhead used by NACK and FEC to VideoSender.
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with  calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder

Note that the logic of how FEC and the needed bitrates is not changed.

BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1972083002 .

Cr-Commit-Position: refs/heads/master@{#13018}
2016-06-02 13:45:53 +00:00
sprang
52033d6ea1 Add H264 bitstream rewriting to limit frame reordering marker in header
The VUI part an SPS may specify max_num_reorder_frames and
max_dec_frame_buffering. These may cause a decoder to buffer a number
of frame prior allowing decode, leading to delays, even if no frames
using such references (ie B-frames) are sent.

Because of this we update any SPS block emitted by the encoder.

Also, a bunch of refactoring of H264-related code to reduce code
duplication.

BUG=

Review-Url: https://codereview.webrtc.org/1979443004
Cr-Commit-Position: refs/heads/master@{#13010}
2016-06-02 09:43:38 +00:00
kjellander
208d19845d Rename APK tests workaround to make it more generic.
We plan to add junit tests running with Robolectric
so naming these files "apk" is slightly confusing.

NOTRY=True

Review-Url: https://codereview.webrtc.org/2020213002
Cr-Commit-Position: refs/heads/master@{#12971}
2016-05-31 11:01:47 +00:00
Peter Boström
cc1543abf3 Move H264BitstreamParser to video_coding.
Moves parser, used in video_coding/ from rtp_rtcp where it is unused.

BUG=webrtc:5678
R=asapersson@webrtc.org
TBR=glaznev@webrt.org

Review URL: https://codereview.webrtc.org/2007553003 .

Cr-Commit-Position: refs/heads/master@{#12866}
2016-05-24 10:16:39 +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
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