708 Commits

Author SHA1 Message Date
solenberg
796b8f9d71 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
2017-03-02 01:02:23 +00:00
ilnik
8dccd67520 In full-stack tests: fixed VP8 simulcast to not decode non-selected streams.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2728553003
Cr-Commit-Position: refs/heads/master@{#16948}
2017-03-01 15:10:30 +00:00
perkj
c5726c1579 Cleanup video test target dependencies in video_test_common.
The purpose is to fix (Asan discovered) duble definitions in upstream project.

BUG=none

Review-Url: https://codereview.webrtc.org/2721303002
Cr-Commit-Position: refs/heads/master@{#16939}
2017-03-01 11:37:08 +00:00
perkj
16ccfdf457 Reland Move fake_audio_device to its own target.
Patchset 1 is patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/
Patchset 2 fix call_perf_test dep on fake_audio_device.

This reverts commit 985371bda999c6db51286586c5850d2ff58f3511.

Original cl description:

Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.

For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.

BUG=none
TBR=kjellander@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2718363002
Cr-Commit-Position: refs/heads/master@{#16922}
2017-02-28 22:41:05 +00:00
sprang
c1b57a15bf Test field trial group with startswith rather than equals.
BUG=webrtc:7266

Review-Url: https://codereview.webrtc.org/2717973005
Cr-Commit-Position: refs/heads/master@{#16915}
2017-02-28 16:50:47 +00:00
philipel
50235b77d0 Make FakeNetworkPipe not busy loop any more.
BUG=webrtc:7259, webrtc:7255

Review-Url: https://codereview.webrtc.org/2718343002
Cr-Commit-Position: refs/heads/master@{#16896}
2017-02-28 10:19:33 +00:00
perkj
985371bda9 Revert of Move fake_audio_device to its own target. (patchset #5 id:80001 of https://codereview.webrtc.org/2717983003/ )
Reason for revert:
Breaks build DEPS.

Original issue's description:
> Move fake_audio_device to its own target.
> The purpose is to make it usefull for test targets that does not need or can use test_common.
>
> For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2717983003
> Cr-Commit-Position: refs/heads/master@{#16889}
> Committed: 03d850ddf9

TBR=ehmaldonado@webrtc.org,danilchap@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none

Review-Url: https://codereview.webrtc.org/2718083003
Cr-Commit-Position: refs/heads/master@{#16890}
2017-02-28 08:56:28 +00:00
perkj
03d850ddf9 Move fake_audio_device to its own target.
The purpose is to make it usefull for test targets that does not need or can use test_common.

For some reason this also triggered override issues in rtp module tests that are fixed in the same cl.

BUG=none

Review-Url: https://codereview.webrtc.org/2717983003
Cr-Commit-Position: refs/heads/master@{#16889}
2017-02-28 08:49:48 +00:00
perkj
a8ba195db5 Replace test::FrameGenerator::ChromaGenerator with new FrameGenerator::SquareGenerator The problem with the ChromaGenerator is that the VP8 encoder produce a too low bitrate for each frame. The squaregenerator make the VP8 encoder produce about 600kbit/s at VGA.
SquareGenerator is a FrameGenerator that draws 10 randomly sized and colored
squares. Between each new generated frame, the squares are moved slightly
towards the lower right corner.

BUG=webrtc:7192

Review-Url: https://codereview.webrtc.org/2705973002
Cr-Commit-Position: refs/heads/master@{#16870}
2017-02-27 14:52:10 +00:00
brandtr
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
kthelgason
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
philipel
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
philipel
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
nisse
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
brandtr
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
ilnik
5328b9eb32 added WebRTC-QuickPerfTest to RampUpTests and CallPerfTests
BUG=webrtc:7153

Review-Url: https://codereview.webrtc.org/2708723002
Cr-Commit-Position: refs/heads/master@{#16743}
2017-02-21 13:20:28 +00:00
stefan
a518a39963 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}
2017-02-21 12:12:23 +00:00
brandtr
872104ac41 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}
2017-02-21 11:59:15 +00:00
nisse
5c29a7aad1 Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket.
Preparing for a media-independent RTP receive stream interface.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2686273002
Cr-Commit-Position: refs/heads/master@{#16646}
2017-02-16 14:52:32 +00:00
solenberg
08b19dfc67 Remove VoEVideoSync interface.
The removed tests are covered by cases in call_perf_tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
2017-02-15 08:42:31 +00:00
solenberg
e374e0139b Remove VoEExternalMedia interface.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2645033002
Cr-Commit-Position: refs/heads/master@{#16608}
2017-02-14 12:55:00 +00:00
brandtr
49ce67c992 Do not encode frames in MultithreadedFakeH264Encoder after Release().
Other minor changes:
- Define locks after stuff it is protecting
- Use explicit default dtors
- Replace unnecessary lock in DelayedEncoder with SequencedTaskChecker

BUG=webrtc:7130

Review-Url: https://codereview.webrtc.org/2686103002
Cr-Commit-Position: refs/heads/master@{#16554}
2017-02-11 08:25:18 +00:00
ehmaldonado
88df0bc591 Make functions in fileutils.h use "const std::string&".
This way, the strings are not copied everytime the function is called.

BUG=webrtc:7142
NOTRY=True

Review-Url: https://codereview.webrtc.org/2685583009
Cr-Commit-Position: refs/heads/master@{#16537}
2017-02-10 17:27:14 +00:00
kwiberg
087bd34d23 Move AudioDecoder and related stuff to the api/ directory
BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 16:15:44 +00:00
stefan
e9ad271db4 Increase the send-time history to 60 seconds.
This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}
2017-02-10 14:09:28 +00:00
perkj
488c5dcd8b Add new target direct_transport and remove fake_network and direct_transport from test_common.
The purpose is to be able to use this feature without building and using all of test_common. However, test_common still builds call_test.cc that is depending on direct_transport.

BUG=none

Review-Url: https://codereview.webrtc.org/2686633002
Cr-Commit-Position: refs/heads/master@{#16494}
2017-02-08 13:55:51 +00:00
stefan
7de8d64f89 Wire up audio packet loss to BWE.
BUG=webtrc:5079

Review-Url: https://codereview.webrtc.org/2658233002
Cr-Commit-Position: refs/heads/master@{#16474}
2017-02-07 15:14:08 +00:00
solenberg
bd9a77f4e5 Remove most of the remaining calls to VoE APIs from Audio[Send|Receive]Stream.
(TBRing webrtc/test/ OWNER)

BUG=webrtc:4690
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2669153004
Cr-Commit-Position: refs/heads/master@{#16455}
2017-02-06 20:53:57 +00:00
stefan
b77c716d8a Enable send-side BWE by default for video in call tests.
Also fixes a bug where RTCP transport feedback was sent even though RTCP was disabled.

May affect perf numbers since the behavior of the send-side BWE differs a lot from the recv-side BWE.

BUG=webrtc:7111

Review-Url: https://codereview.webrtc.org/2669413003
Cr-Commit-Position: refs/heads/master@{#16451}
2017-02-06 14:29:38 +00:00
ehmaldonado
656610fbe7 Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
Remove video_capture as a dependency of test_common and add it as a dependency of modules_unittests, as it was before the refactor in https://codereview.webrtc.org/2629923002

BUG=webrtc:7037
NOTRY=True

Review-Url: https://codereview.webrtc.org/2666113003
Cr-Commit-Position: refs/heads/master@{#16439}
2017-02-06 10:21:11 +00:00
sprang
b1ca073db4 Rename adaptation api methods, extended vie_encoder unit test.
Use AdaptDown/AdaptUp instead of ScaleDown/ScaleUp, since we may want to
adapt using other means than resolution.

Also, extend vie_encoder with unit test that actually uses frames scaled
to resolution as determined by VideoAdapter, since that seems to be the
default implementation.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2652893015
Cr-Commit-Position: refs/heads/master@{#16402}
2017-02-01 16:38:12 +00:00
elad.alon
d83b9670a6 Replace consecutive-losses count by a calculation of first-order-FEC recoverability.
Note:
* PLR is calculated over all of the known packets.
* RPLR is calculated over all of the known packet *pairs*. That is, only over sets of subsequent packets where the reception status is known for both.

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2629883003
Cr-Commit-Position: refs/heads/master@{#16401}
2017-02-01 16:36:09 +00:00
perkj
ac61b745df Refactor FakeAudioDevice to have separate methods for starting recording and playout.
Also, change FakeAudioDevice to generate a sine tone instead of using a file.

TBR=henrika@webrtc.org, stefan@webrtc.org

BUG=webrtc:7080

Review-Url: https://codereview.webrtc.org/2652803002
Cr-Commit-Position: refs/heads/master@{#16385}
2017-01-31 21:32:49 +00:00
solenberg
3ebbcb528b Stop using VoEVideoSync in Call/VideoReceiveStream.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2452163004
Cr-Commit-Position: refs/heads/master@{#16375}
2017-01-31 11:58:40 +00:00
nisse
1c0dea8675 Delete VideoFrame::set_render_time_ms.
Also mark the render_time_ms getter function and the ntp timestamp
as deprecated.

BUG=webrtc:6977

Review-Url: https://codereview.webrtc.org/2633493002
Cr-Commit-Position: refs/heads/master@{#16354}
2017-01-30 10:43:18 +00:00
brandtr
fb45c6c103 Inform jitter buffer about FlexFEC protection.
This CL introduces a way for the VideoReceiveStreams to check whether
they are protected by any FlexfecReceiveStreams. This is done in the
VideoReceiveStream::Start() method, which then propagates this information
down to the jitter buffer adaptation logic.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2649973005
Cr-Commit-Position: refs/heads/master@{#16328}
2017-01-27 14:47:55 +00:00
mbonadei
9aa3f0a200 Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reason for revert:
Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file)

Original issue's description:
> Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
>
> Reason for revert:
> This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio
>
> Original issue's description:
> > Moving webrtc.gni up one level from build/
> >
> > BUG=webrtc:7030
> >
> > Review-Url: https://codereview.webrtc.org/2651543003
> > Cr-Commit-Position: refs/heads/master@{#16241}
> > Committed: 35a32700fc
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2657563002
> Cr-Commit-Position: refs/heads/master@{#16244}
> Committed: 69dc7dbe24

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2654773002
Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 14:58:22 +00:00
mbonadei
69dc7dbe24 Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ )
Reason for revert:
This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio

Original issue's description:
> Moving webrtc.gni up one level from build/
>
> BUG=webrtc:7030
>
> Review-Url: https://codereview.webrtc.org/2651543003
> Cr-Commit-Position: refs/heads/master@{#16241}
> Committed: 35a32700fc

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2657563002
Cr-Commit-Position: refs/heads/master@{#16244}
2017-01-24 13:14:35 +00:00
mbonadei
35a32700fc Moving webrtc.gni up one level from build/
BUG=webrtc:7030

Review-Url: https://codereview.webrtc.org/2651543003
Cr-Commit-Position: refs/heads/master@{#16241}
2017-01-24 12:49:35 +00:00
minyue
435ddf978d Add TransportFeedbackPacketLossTracker.
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.

BUG=webrtc:6904

Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
2017-01-23 16:07:05 +00:00
ossu
7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.

Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.

Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.

BUG=webrtc:5883

Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 12:56:25 +00:00
ehmaldonado
3626865be2 GN: Refactor modules_unittests to eliminate package boundary violations.
BUG=webrtc:6954

Review-Url: https://codereview.webrtc.org/2629923002
Cr-Commit-Position: refs/heads/master@{#16166}
2017-01-19 16:27:11 +00:00
brandtr
fa5a368b3c Let FlexfecReceiveStreamImpl send RTCP RRs.
This CL adds an RTP module to FlexfecReceiveStreamImpl, and wires it up
to send RTCP RRs. It further makes some methods take const refs instead
of values, to make it more clear where packet copies are made. This
change reduces the number of copies by one, for the case when media
packets are added to the FlexFEC receiver.

The end-to-end test is modified to check for RTCP RRs being sent.
Part of this modification involves some indentation changes, and the
diff thus looks bigger than it logically is.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2625633003
Cr-Commit-Position: refs/heads/master@{#16106}
2017-01-17 09:33:54 +00:00
brandtr
3d200bd6ac Remove FlexfecConfig and replace with specific struct in VideoSendStream.
The existence of FlexfecConfig is due to a naive design. Now when it
is not used on the receiving side (see https://codereview.webrtc.org/2542413002),
it is time to remove it from the sending side as well.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2621573002
Cr-Commit-Position: refs/heads/master@{#16097}
2017-01-16 14:59:19 +00:00
brandtr
e78d26669e Make FakeEncoder and FakeH264Encoder thread safe.
The MultithreadedFakeH264Encoder is a derived class from FakeEncoder
and FakeH264Encoder, and these should thus also be thread safe.

TESTED=Ran "out/Tsan/video_engine_tests --gtest_filter="*Multithreaded*" --gtest_repeat=100" with is_debug=false, dcheck_always_on=true, is_tsan=true.

BUG=webrtc:6943

Review-Url: https://codereview.webrtc.org/2604403003
Cr-Commit-Position: refs/heads/master@{#16093}
2017-01-16 13:57:16 +00:00
brandtr
8313a6fa8f Make |rtcp_send_transport| mandatory in FlexfecReceiveStream::Config.
That object will be used when we enable RTCP reporting from FlexfecReceiveStream.

Other related changes:
- Stop using FlexfecConfig (from config.h) at receive side in WebRtcVideoEngine2.
- Add a IsCompleteAndEnabled() method to FlexfecReceiveStream::Config, to be
  used in WebRtcVideoEngine2.
- Centralize the construction of the FlexfecReceiveStream::Config in unit tests.
  This will make future additions to the unit tests cleaner.
- Simplify setup for receiving FlexFEC in VideoQualityTest.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2589713003
Cr-Commit-Position: refs/heads/master@{#16059}
2017-01-13 15:41:19 +00:00
philipel
ea142f8de3 Don't detect a new frame if a previous packet is used in a previous frame.
In this CL:
 - Removed unused variable |last_seq_num_|.
 - Fixed bug where a new incomplete frame was detected as a complete frame.
 - Added fuzzer to video_coding::PacketBuffer.

BUG=chromium:677101

Review-Url: https://codereview.webrtc.org/2613833003
Cr-Commit-Position: refs/heads/master@{#16003}
2017-01-11 10:01:56 +00:00
nisse
af916899cc Move VideoFrame and related declarations to webrtc/api/video.
Moves webrtc/common_video/rotation.h and parts of
webrtc/common_video/include/video_frame_buffer.h and
webrtc/video_frame.h, and adds to a new GN target api:video_frame_api.

BUG=webrtc:5880

Review-Url: https://codereview.webrtc.org/2517173004
Cr-Commit-Position: refs/heads/master@{#15993}
2017-01-10 15:44:26 +00:00
kthelgason
5d682ca7a1 Disable some filesystem tests that don't make sense on iOS.
BUG=webrtc:5571
NOTRY=true

Review-Url: https://codereview.webrtc.org/2627583002
Cr-Commit-Position: refs/heads/master@{#15981}
2017-01-10 11:00:41 +00:00
kthelgason
c1b661eee0 Remove duplicate implementations of string conversion methods.
BUG=webrtc:4773

Review-Url: https://codereview.webrtc.org/2583853002
Cr-Commit-Position: refs/heads/master@{#15957}
2017-01-09 09:40:03 +00:00