Commit Graph

9819 Commits

Author SHA1 Message Date
8697f9afba Skips the first frame in DxgiDuplicatorController
A bug has been reported to complaint the ScreenCapturerWinDirectx cannot capture
the first frame, which is used in "Report an issue" page.
A simple solution here is to skip the first frame.

This change also removes the friend relationship between
DxgiDuplicatorController / DxgiAdapterDuplicator / DxgiOutputDuplicator, which
is not really necessary.

BUG=682112

Review-Url: https://codereview.webrtc.org/2703123002
Cr-Commit-Position: refs/heads/master@{#16815}
2017-02-24 05:43:40 +00:00
162cb53e7b Making AsyncInvoker destructor thread-safe.
The documentation for AsyncInvoker states that it owns the lifetime of
calls, and when its destructor is called, all in-flight calls are
cancelled or finish executing. The "cancelled" part is working, but if
a call is in the middle of executing, the destructor does *not* wait.

This is fixed by keeping a count of pending invocations, which is
decremented when a call is either cleared from a message queue or
finishes executing.

BUG=webrtc:3914, webrtc:3911

Review-Url: https://codereview.webrtc.org/2694723004
Cr-Commit-Position: refs/heads/master@{#16811}
2017-02-24 01:10:07 +00:00
0d1305ee88 Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533
This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented.
This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume

BUG=webrtc:6533, webrtc:6805

This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong.

Review-Url: https://codereview.webrtc.org/2534843002
Cr-Commit-Position: refs/heads/master@{#16809}
2017-02-23 21:57:00 +00:00
384498abb5 Use sched_yield on all POSIX platforms in PlatformThread.
(not only MacOS)

This is a test to see if perf regressions we're seeing may be related to the use of nanosleep().

BUG=695438
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2716683002 .
Cr-Commit-Position: refs/heads/master@{#16807}
2017-02-23 19:23:19 +00:00
0e22a4cfd3 Android HW decoder: Support odd heights for non-texture output
When textures are not enabled and we are using byte buffer outputs, the
decoder is currently crashing for odd heights because of an RTC_CHECK.
This CL removes the check and handles the pointer offset to the chroma
planes for the odd height case instead.

This has been verified to work correctly on a Pixel device.

BUG=webrtc:6651

Review-Url: https://codereview.webrtc.org/2709923005
Cr-Commit-Position: refs/heads/master@{#16805}
2017-02-23 15:11:32 +00:00
88b82b59f1 Fix compilation issue detected by internal tool.
Explicit initialization of const member of new EchoCanceller 3
submodule.

Same thing as f3e84717 from https://codereview.webrtc.org/2715573003,
but more of it.

This should be the last one.

TBR=henrik.lundin@webrtc.org

NOTRY=True
BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2708373005
Cr-Commit-Position: refs/heads/master@{#16804}
2017-02-23 14:27:03 +00:00
f3e847170d Fix compilation issue detected by internal tool.
Explicit initialization of const member of new EchoCanceller 3
submodule.

NOTRY=True

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2715573003
Cr-Commit-Position: refs/heads/master@{#16802}
2017-02-23 13:56:15 +00:00
522d71bf36 Finalization of the first version of EchoCanceller 3
This CL adds the remaining code for the first version of EchoCanceller3.

TBR=aleloi@webrtc.org
BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2678423005
Cr-Commit-Position: refs/heads/master@{#16801}
2017-02-23 13:16:26 +00:00
1e7732c3d9 Fixed Full stack tests to correctly process selected TL and SL while
calculating frame sizes. Added actual_bitrate metric which also accounts
for TL and SL info. Metric encoded_frame_size calculation is cleaned up. Perf alerts should be ignored.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2709483009
Cr-Commit-Position: refs/heads/master@{#16800}
2017-02-23 13:07:56 +00:00
560ddb7321 Set scaling limit at 320 * 180 for all implementations.
The MediaCodec decoder on android has trouble decoding video at
so low resolutions. We set the limit a bit higher for all implementations
pending a robust software fallback implementation for MediaCodec.

BUG=webrtc:7206

Review-Url: https://codereview.webrtc.org/2709153002
Cr-Commit-Position: refs/heads/master@{#16798}
2017-02-23 11:22:02 +00:00
8aadd50b96 Propagate packet pacing information to SendTimeHistory.
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/

webrtc::PacedSender::Process                        <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket         <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld            <--- this CL end here

BUG=webrtc:6822

Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
2017-02-23 10:56:13 +00:00
7adadb1fdf Add QP for libvpx VP9 decoder.
Also adds a basic unit test for VP9 implementation.

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2654813002
Cr-Commit-Position: refs/heads/master@{#16795}
2017-02-23 10:54:57 +00:00
1e46434e7e Rename config structs in VideoProcessor tests.
This CL harmonizes and improves the naming of the config structs
in the VideoProcessor tests. It should have no functional implications.

CodecConfigPars -> CodecParams:
  This struct mainly contains codec settings.

QualityMetrics -> QualityThresholds:
  This struct contains thresholds against which the calculated
  PSNR and SSIM metrics are compared to.

RateControlMetrics -> RateControlThresholds:
  This struct contains thresholds against which the calculated
  rate control metrics are compared to.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2703333004
Cr-Commit-Position: refs/heads/master@{#16794}
2017-02-23 10:49:16 +00:00
f166e1bcab Now using rtc::Buffer in FineAudioBuffer
BUG=b/35589717

Review-Url: https://codereview.webrtc.org/2706923006
Cr-Commit-Position: refs/heads/master@{#16793}
2017-02-23 10:44:55 +00:00
02f994b4e9 Remove codec thread from MediaCodecVideoEncoder.
After this change, all calls to MediaCodecVideoEncoder must be made on
the same task queue. Removes OnCodecThread suffix from methods since it
is no longer meaningful.

BUG=webrtc:6290

Review-Url: https://codereview.webrtc.org/2669093004
Cr-Commit-Position: refs/heads/master@{#16792}
2017-02-23 10:25:20 +00:00
df92c5cb8c Adding cpu measurments to video_quality_tests
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2711493002
Cr-Commit-Position: refs/heads/master@{#16791}
2017-02-23 10:08:44 +00:00
abc0080df8 Add QP statistics to VideoProcessorIntegrationTest.
The average QP of encoded frames is printed in Stats::PrintSummary.

plot_webrtc_test_logs.py: Add QP to plots.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2709613005
Cr-Commit-Position: refs/heads/master@{#16790}
2017-02-23 09:33:04 +00:00
61a2b1bd6c Micro change suggested by internal style tool.
BUG=None
TBR=philipel@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2707973009
Cr-Commit-Position: refs/heads/master@{#16789}
2017-02-23 09:16:14 +00:00
b681aabdfc Revert of Add metal view, shaders and renderer. (patchset #18 id:340001 of https://codereview.webrtc.org/2651743007/ )
Reason for revert:
Reverting due to breakage in the Google3 import

Original issue's description:
> Add metal view, shaders and renderer.
>
> This CL submits standalone Metal view, renderer and shader.
>
> BUG=webrtc:7079
>
> Review-Url: https://codereview.webrtc.org/2651743007
> Cr-Commit-Position: refs/heads/master@{#16787}
> Committed: fc8c97f950

TBR=magjed@webrtc.org,kthelgason@webrtc.org,tkchin@webrtc.org,haysc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7079

Review-Url: https://codereview.webrtc.org/2711003004
Cr-Commit-Position: refs/heads/master@{#16788}
2017-02-23 09:15:07 +00:00
fc8c97f950 Add metal view, shaders and renderer.
This CL submits standalone Metal view, renderer and shader.

BUG=webrtc:7079

Review-Url: https://codereview.webrtc.org/2651743007
Cr-Commit-Position: refs/heads/master@{#16787}
2017-02-23 08:46:07 +00:00
8c80c6e389 Fix potential deadlock in TaskQueue's libevent PostTaskAndReply implementation
BUG=webrtc:7188

Review-Url: https://codereview.webrtc.org/2709603002
Cr-Commit-Position: refs/heads/master@{#16786}
2017-02-23 08:34:52 +00:00
4b1bf6c2f0 Adding placeholder ortc_unittests target.
This will allow the trybots to be updated to start running this new test
executable, so that they can be used when landing this CL which will
replace the dummy test with real tests:
https://codereview.webrtc.org/2675173003/

BUG=webrtc:7013
TBR=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2707013005
Cr-Commit-Position: refs/heads/master@{#16784}
2017-02-23 07:45:38 +00:00
b789253661 Accept SDP with TRANSPORT attributes missing from bundled m= sections.
Where "TRANSPORT attributes" refers to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-16

The BUNDLE draft now says that these attributes can
(in fact, MUST) be omitted when m= sections are bundled
(they only need to go in one of the bundled m= sections),
so we should start accepting that SDP.

This CL doesn't fix "a=rtcp-mux", unfortunately. That will be easier
to fix once we've split apart an "RtpTransport" object from
BaseChannel.

BUG=webrtc:6351

Review-Url: https://codereview.webrtc.org/2647593003
Cr-Commit-Position: refs/heads/master@{#16782}
2017-02-23 03:35:18 +00:00
3fa87f782e Use FallbackDesktopCapturerWrapper in ScreenCapturerWinMagnifier
This is a trivial change to remove duplicate logic, i.e. fallback capturer, from
ScreenCapturerWinMagnifier.

BUG=webrtc:7215

Review-Url: https://codereview.webrtc.org/2704943002
Cr-Commit-Position: refs/heads/master@{#16781}
2017-02-22 21:47:00 +00:00
0f8b403eb5 Introduce a new constructor to PlatformThread.
The new constructor introduces two new changes:

* Support specifying thread priority at construction time.
  - Moving forward, the SetPriority() method will be removed.
* New thread function type.
  - The new type has 'void' as a return type and a polling loop
    inside PlatformThread, is not used.

The old function type is still supported until all places have been moved over.

In this CL, the first steps towards deprecating the old mechanism are taken
by moving parts of the code that were simple to move, over to the new callback
type.

BUG=webrtc:7187

Review-Url: https://codereview.webrtc.org/2708723003
Cr-Commit-Position: refs/heads/master@{#16779}
2017-02-22 19:22:05 +00:00
6d230d7b1d Fix issue where video scaling gets stuck at low resolution
This CL fixes issue 7211 as well as adding a test that would have
caught the issue.

BUG=webrtc:7211,webrtc:6850,b/35471214
R=sprang@webrtc.org
TBR=kthelgason, sprang

Review-Url: https://codereview.webrtc.org/2713683002 .
Cr-Commit-Position: refs/heads/master@{#16778}
2017-02-22 17:30:27 +00:00
6f08d7d68d Change frame length on very low bandwidth.
BUG=webrtc:7199

Review-Url: https://codereview.webrtc.org/2703353002
Cr-Commit-Position: refs/heads/master@{#16777}
2017-02-22 15:35:05 +00:00
6e5b2195d7 Add ana config to event log visualiser
BUG=webrtc:7160

Review-Url: https://codereview.webrtc.org/2695613005
Cr-Commit-Position: refs/heads/master@{#16776}
2017-02-22 15:33:27 +00:00
0335e6c4bf Fix flaky test WebRtcMediaRecorderTest.PeerConnection
A previous CL changed from logging to DCHECKing when setting minimum playout delay on a VoE channel: https://codereview.webrtc.org/2452163004/

I thought it safe at the time, since the input parameter range is capped, but apparently I didn't dig deep enough, as ultimately a failure may be returned for other reasons: https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_coding/neteq/delay_manager.cc#381

Thus, reverting to old behavior.

BUG=694373

Review-Url: https://codereview.webrtc.org/2704933008
Cr-Commit-Position: refs/heads/master@{#16775}
2017-02-22 15:07:04 +00:00
1d4e3d8a2e Move rtc_task_runner dependency from rtc_base to rtc_base_unittests.
This is step 3 in the task runner migration process started in cl
https://codereview.webrtc.org/2696703009/.

It depends on step 2 being landed in Chrome, cl
https://codereview.chromium.org/2694363005/.

NOTRY=true
BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2708843002
Cr-Commit-Position: refs/heads/master@{#16774}
2017-02-22 14:02:34 +00:00
a45102f7b4 Revert of Revert Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2682073003/ )
Reason for revert:
Fix here: https://codereview.chromium.org/2708593003

Original issue's description:
> Revert Make the new jitter buffer the default jitter buffer.
>
> Speculative revert of https://codereview.chromium.org/2656983002/ to see if it fixes a downstream bug.
>
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2682073003
> Cr-Commit-Position: refs/heads/master@{#16492}
> Committed: e525d6aba6

TBR=nisse@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2704183002
Cr-Commit-Position: refs/heads/master@{#16772}
2017-02-22 13:30:39 +00:00
5650a7d1c4 Improved readability and DCHECKing in AudioVector::[]
This is a follow-up to https://codereview.webrtc.org/2700633003, where
post-commit comments suggested these changes.

BUG=webrtc:7159

Review-Url: https://codereview.webrtc.org/2706263002
Cr-Commit-Position: refs/heads/master@{#16771}
2017-02-22 11:45:40 +00:00
b78bc75e8c Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ )
Reason for revert:
Necessary calls were "protected" by RTC_DCHECKs, that were optimized away in some release builds.
Replacing the RTC_DCHECKs with EXPECTs.

Original issue's description:
> Revert of Add optional visualization file writers to VideoProcessor tests. (patchset #4 id:220001 of https://codereview.webrtc.org/2700493006/ )
>
> Reason for revert:
> Breaks downstream project.
>
> Original issue's description:
> > Add optional visualization file writers to VideoProcessor tests.
> >
> > The purpose of this visualization CL is to add the ability to record
> > video at the source, after encode, and after decode, in the VideoProcessor
> > tests. These output files can then be replayed and used as a subjective
> > complement to the objective metric plots given by the existing Python
> > plotting script.
> >
> > BUG=webrtc:6634
> >
> > Review-Url: https://codereview.webrtc.org/2700493006
> > Cr-Commit-Position: refs/heads/master@{#16738}
> > Committed: 872104ac41
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6634
>
> Review-Url: https://codereview.webrtc.org/2708103002
> Cr-Commit-Position: refs/heads/master@{#16745}
> Committed: 2a8135a174

TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2706123003
Cr-Commit-Position: refs/heads/master@{#16769}
2017-02-22 09:26:59 +00:00
798781299f Count FlexFEC packets in Call UMA stats.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2684243002
Cr-Commit-Position: refs/heads/master@{#16768}
2017-02-22 09:20:01 +00:00
1e5b0269a8 Updating system_wrappers/include/metrics.h docs
In the metrics.h documentation the target to include a default
implementation of metrics was referring to the previous build system
(gyp). Now it is updated to refer to the current target.

BUG=None
NOTRY=True
TBR=henrika@webrtc.org

Review-Url: https://codereview.webrtc.org/2699093002
Cr-Commit-Position: refs/heads/master@{#16766}
2017-02-22 08:55:32 +00:00
de6adbe0ba Remove non-ARC code from the codebase.
BUG=webrtc:7198

Review-Url: https://codereview.webrtc.org/2702153004
Cr-Commit-Position: refs/heads/master@{#16765}
2017-02-22 08:42:11 +00:00
59fc9030ea Remove codec setting members in VideoProcessorIntegrationTest. Use settings in CodecConfigPars directly instead.
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2707763005
Cr-Commit-Position: refs/heads/master@{#16764}
2017-02-22 07:47:53 +00:00
9ffa13ff5d Don't attempt to use "network binder" for sockets bound to "ANY" IP.
BUG=NONE

Review-Url: https://codereview.webrtc.org/2701253002
Cr-Commit-Position: refs/heads/master@{#16760}
2017-02-22 00:18:00 +00:00
e352dbe6d5 Update comments in FallbackDesktopCapturerWrapper
Update the year in copyright headers from 2016 to 2017, and also rename a
variable in FallbackDesktopCapturerWrapperTest to follow coding style.

BUG=webrtc:7205

Review-Url: https://codereview.webrtc.org/2706193005
Cr-Commit-Position: refs/heads/master@{#16759}
2017-02-21 23:00:07 +00:00
996103a19f Make use_single_core option configurable in VideoProcessorIntegrationTests.
plot_webrtc_test_logs.py: Add number of used cores to figure title.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2706753005
Cr-Commit-Position: refs/heads/master@{#16756}
2017-02-21 16:30:04 +00:00
087613c8df Rename AudioMixer factory method.
AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(rate_calculator, bool limiter)

was added to create a mixer without the limiter subcomponent. Calling
it "Create with ... *and* limiter" is counterintuitive.

Renamed to simply 'Create'.

TBR=solenberg@webrtc.org

BUG=webrtc:7167

Review-Url: https://codereview.webrtc.org/2709523006
Cr-Commit-Position: refs/heads/master@{#16755}
2017-02-21 16:27:08 +00:00
6f142eb36e Add protection for RTCPSender::max_packet_size_.
This cl protects the access to the max_packet_size_, without fixing
the underlying race; the value is simply copied to a local variable,
whose value might be stale when used.

BUG=webrtc:7189

Review-Url: https://codereview.webrtc.org/2704263003
Cr-Commit-Position: refs/heads/master@{#16754}
2017-02-21 15:32:47 +00:00
5ef2bc1914 Reland of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #1 id:1 of https://codereview.chromium.org/2703393002/ )
Reason for revert:
Downstream fixed

Original issue's description:
> Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ )
>
> Reason for revert:
> Breaks downstream
>
> Original issue's description:
> > Fixes a bug where a video stream can get stuck in the suspended state.
> >
> > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough.
> >
> > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before.
> >
> > BUG=webrtc:7178
> >
> > Review-Url: https://codereview.webrtc.org/2705603002
> > Cr-Commit-Position: refs/heads/master@{#16739}
> > Committed: a518a39963
>
> TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7178
>
> Review-Url: https://codereview.webrtc.org/2703393002
> Cr-Commit-Position: refs/heads/master@{#16751}
> Committed: b80bdcafed

TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7178

Review-Url: https://codereview.webrtc.org/2704323003
Cr-Commit-Position: refs/heads/master@{#16753}
2017-02-21 15:28:31 +00:00
b80bdcafed Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ )
Reason for revert:
Breaks downstream

Original issue's description:
> Fixes a bug where a video stream can get stuck in the suspended state.
>
> This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough.
>
> This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before.
>
> BUG=webrtc:7178
>
> Review-Url: https://codereview.webrtc.org/2705603002
> Cr-Commit-Position: refs/heads/master@{#16739}
> Committed: a518a39963

TBR=mflodman@webrtc.org,terelius@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7178

Review-Url: https://codereview.webrtc.org/2703393002
Cr-Commit-Position: refs/heads/master@{#16751}
2017-02-21 14:52:26 +00:00
657bab2455 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
This avoids redoing RTP header parsing already done in Call.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2697833002
Cr-Commit-Position: refs/heads/master@{#16750}
2017-02-21 14:28:10 +00:00
b94491d790 Implement operator<< for AudioCodec
It's annoying to have to re-implement this every time I need a debug
printout.

Declared inline, so that there'll be zero runtime overhead.

This CL also modifies a unit test so that it will make use of the new
operator<< in case it finds errors.

BUG=none

Review-Url: https://codereview.webrtc.org/2705203002
Cr-Commit-Position: refs/heads/master@{#16749}
2017-02-21 14:16:19 +00:00
ec067e9d21 Reduce usage of tmmbr information structure
by creating it on accepted tmmbr/tmmbn rtcp messages
rather on sender/receiver reports.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2702373002
Cr-Commit-Position: refs/heads/master@{#16748}
2017-02-21 13:38:19 +00:00
4e4dfbd45d Move YuvConverter from Android API to src.
BUG=webrtc:7172

Review-Url: https://codereview.webrtc.org/2705173002
Cr-Commit-Position: refs/heads/master@{#16747}
2017-02-21 13:29:59 +00:00
c3c46246a9 Add RTCVideoFrame init function from CVPixelBufferRef
Adds a public init function in RTCVideoFrame that makes it possible to
create a frame from a CVPixelBufferRef.

BUG=webrtc:7177
NOTRY=True

Review-Url: https://codereview.webrtc.org/2700113003
Cr-Commit-Position: refs/heads/master@{#16746}
2017-02-21 13:28:48 +00:00
2a8135a174 Revert of Add optional visualization file writers to VideoProcessor tests. (patchset #4 id:220001 of https://codereview.webrtc.org/2700493006/ )
Reason for revert:
Breaks downstream project.

Original issue's description:
> Add optional visualization file writers to VideoProcessor tests.
>
> The purpose of this visualization CL is to add the ability to record
> video at the source, after encode, and after decode, in the VideoProcessor
> tests. These output files can then be replayed and used as a subjective
> complement to the objective metric plots given by the existing Python
> plotting script.
>
> BUG=webrtc:6634
>
> Review-Url: https://codereview.webrtc.org/2700493006
> Cr-Commit-Position: refs/heads/master@{#16738}
> Committed: 872104ac41

TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2708103002
Cr-Commit-Position: refs/heads/master@{#16745}
2017-02-21 13:24:03 +00:00