9806 Commits

Author SHA1 Message Date
henrika
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
sakal
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
ilnik
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
asapersson
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
aleloi
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
denicija
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
denicija
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
tommi
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
deadbeef
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
deadbeef
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
zijiehe
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
tommi
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
Magnus Jedvert
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
michaelt
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
michaelt
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
solenberg
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
nisse
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
philipel
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
henrik.lundin
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
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
brandtr
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
mbonadei
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
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
asapersson
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
deadbeef
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
zijiehe
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
asapersson
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
aleloi
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
nisse
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
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
kwiberg
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
danilchap
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
sakal
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
magjed
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
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
Tommi
5dd5f5a319 RembWithSendSideBwe: Rename |event_| to |stop_event_| and set it when the test ends.
BUG=webrtc:7200
R=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2706223002 .
Cr-Commit-Position: refs/heads/master@{#16744}
2017-02-21 13:22:59 +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
aleloi
24899e58ec Optionally disable APM limiter in AudioMixer.
The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping).

To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner.

The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter

Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other.

This also fixes a few minor GN issues so that warnings do not have to be suppressed.

BUG=webrtc:7167

Review-Url: https://codereview.webrtc.org/2692333002
Cr-Commit-Position: refs/heads/master@{#16742}
2017-02-21 13:06:29 +00:00
magjed
7ee512581c Clean up RTCVideoFrame
RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame, but it
currently contains some extra logic beyond that. We want RTCVideoFrame
to be as simple as possible, i.e. just a container with no extra state,
so we can use it as input to RTCVideoSource without complicating the
interface for consumers.

BUG=webrtc:7177
NOTRY=True

Review-Url: https://codereview.webrtc.org/2695203004
Cr-Commit-Position: refs/heads/master@{#16740}
2017-02-21 12:19:46 +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
7d59f6b1c4 Reland of Delete class SSRCDatabase, and its global ssrc registry. (patchset #1 id:1 of https://codereview.webrtc.org/2700413002/ )
Reason for revert:
Intend to fix perf problem and reland.

Original issue's description:
> Revert of Delete class SSRCDatabase, and its global ssrc registry. (patchset #20 id:370001 of https://codereview.webrtc.org/2644303002/ )
>
> Reason for revert:
> Breaks webrtc_perf_tests reliably:
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/1780
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus4%29/builds/178
>
> We're actively working on getting a quick version of webrtc_perf_tests up on the trybots again to prevent breakages like this: https://bugs.chromium.org/p/webrtc/issues/detail?id=7101
>
> Original issue's description:
> > Delete class SSRCDatabase, and its global ssrc registry,
> > and the method RTPSender::GenerateNewSSRC.
> >
> > It's now mandatory for higher layers to call SetSSRC, RTPSender
> > no longer allocates any ssrc by default.
> >
> > BUG=webrtc:4306,webrtc:6887
> >
> > Review-Url: https://codereview.webrtc.org/2644303002
> > Cr-Commit-Position: refs/heads/master@{#16670}
> > Committed: b78d4d1383
>
> TBR=solenberg@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,ivoc@webrtc.org,nisse@webrtc.org
> NOTRY=True
> BUG=webrtc:4306,webrtc:6887
>
> Review-Url: https://codereview.webrtc.org/2700413002
> Cr-Commit-Position: refs/heads/master@{#16693}
> Committed: b5848ecbf5

TBR=solenberg@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,ivoc@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4306,webrtc:6887

Review-Url: https://codereview.webrtc.org/2702203002
Cr-Commit-Position: refs/heads/master@{#16737}
2017-02-21 11:40:24 +00:00
ilnik
531100dc7a Reland of Added GetCpuTime to base/ to get total CPU time consumed by process for perf tests.
BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2695743003
Cr-Commit-Position: refs/heads/master@{#16665}
Committed: 3ff474b72b

patch from issue 2695743003 at patchset 440001 (http://crrev.com/2695743003#ps440001)

Review-Url: https://codereview.webrtc.org/2706823002
Cr-Commit-Position: refs/heads/master@{#16736}
2017-02-21 11:33:24 +00:00
philipel
e6f1601d08 Revert of Added kNotAProbe definiton to PacketInfo. (patchset #1 id:1 of https://codereview.chromium.org/2697383004/ )
Reason for revert:
Downstream fix landed.

Original issue's description:
> Added kNotAProbe definiton to PacketInfo.
>
> BUG=none
> NOTRY=True
> TBR=nisse@webrtc.org, stefan@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2697383004
> Cr-Commit-Position: refs/heads/master@{#16668}
> Committed: 4db68e609b

TBR=nisse@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=none

Review-Url: https://codereview.webrtc.org/2706823003
Cr-Commit-Position: refs/heads/master@{#16735}
2017-02-21 09:28:41 +00:00
solenberg
76377c55b7 Remove usage of VoEAudioProcessing from WVoE/MC.
Calling APM and TransmitMixer directly instead.

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

Review-Url: https://codereview.webrtc.org/2681033010
Cr-Commit-Position: refs/heads/master@{#16734}
2017-02-21 08:54:31 +00:00
brandtr
11c9eafc69 Build plot_videoprocessor_integrationtest by default.
NOTRY=True
BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2702333002
Cr-Commit-Position: refs/heads/master@{#16733}
2017-02-21 07:56:39 +00:00
nisse
1e32122168 Delete VideoCaptureCapability::codecType and related logic.
The video_capture module includes remnants of support for cameras
producing encoded frames. However, this seems to be unused, and is
explicitly not supported by VideoCaptureImpl::IncomingFrame.

BUG=None

Review-Url: https://codereview.webrtc.org/2668693008
Cr-Commit-Position: refs/heads/master@{#16732}
2017-02-21 07:27:37 +00:00
deadbeef
4024b9bbe6 Move filerotatingstream_unittest.cc to rtc_base_nonparallel_tests.
These tests involve interactions with the file system, so to avoid
flakiness they shouldn't be run in parallel.

BUG=webrtc:7195
NOTRY=True

Review-Url: https://codereview.webrtc.org/2710433003
Cr-Commit-Position: refs/heads/master@{#16727}
2017-02-20 20:07:50 +00:00