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}
Video modules are added in reverse order to ensure that the padding order is the same as before, prioritizing high resolution streams.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2655033002
Cr-Commit-Position: refs/heads/master@{#16329}
This avoids issues where the bitrate produced by the codec is far lower than the target bitrate in the beginning, which causes the delay-based BWE to be initialized accordingly.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2653883002
Cr-Commit-Position: refs/heads/master@{#16327}
This CL adds code to the BlockProcessor, which basically constitutes
the second layer in echo canceller 3. The CL includes two incomplete
classes (EchoRemover and EchoPathDelayEstimator) which will be completed
in upcoming CLs. Because of this, some of the unittests are disabled
until those are added.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2611223003
Cr-Commit-Position: refs/heads/master@{#16319}
This means that smaller probe packets will be allowed at lower bitrates.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2650393002
Cr-Commit-Position: refs/heads/master@{#16317}
The original CL was reverted because of a bug discovered by the
chromium bots. Description of that CL:
> Review-Url: https://codereview.webrtc.org/2636443002
> Cr-Commit-Position: refs/heads/master@{#16135}
> Committed: a28e971e3b
The first patch set of this CL is the same as r16135.
Subsequence patch sets are the fixes applied.
Some new test cases have been added, which reveal a few more bugs that
have also been fixed.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2641133002
Cr-Commit-Position: refs/heads/master@{#16299}
Reason for revert:
Bugfixes related to the new jitter buffer has landed.
Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
>
> Reason for revert:
> Breaks tests downstream.
>
> Original issue's description:
> > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> >
> > Reason for revert:
> > Fix in this CL: https://codereview.chromium.org/2640793003/
> >
> > Original issue's description:
> > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > >
> > > Reason for revert:
> > > Breaks android bots.
> > >
> > > Original issue's description:
> > > > Make the new jitter buffer the default jitter buffer.
> > > >
> > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > buffer, clean up will be done in follow up CLs.
> > > >
> > > > In this CL:
> > > > - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > > new video jitter buffer the default one.
> > > > - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > > WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > >
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > Committed: 0f0763d86d
> > >
> > > TBR=stefan@webrtc.org,terelius@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/2632123005
> > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > Committed: c08c191f7d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2642753002
> > Cr-Commit-Position: refs/heads/master@{#16149}
> > Committed: f20dd0014d
>
> TBR=stefan@webrtc.org,terelius@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/2638423003
> Cr-Commit-Position: refs/heads/master@{#16159}
> Committed: 04926b8264TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2652043005
Cr-Commit-Position: refs/heads/master@{#16293}
Reason for revert:
This was false alarm. The next Chromium roll fixed the problem and the bot has been green for 3 builds.
Original issue's description:
> Disabled two iOS tests due to bot breakage. Affected tests are
> AudioDeviceTest.RunPlayoutWithFileAsSource and
> AudioDeviceTest.StartStopRecording
>
> NOTRY=True
> TBR=henrika@webrtc.org
> BUG=7056
>
> Review-Url: https://codereview.webrtc.org/2652423002
> Cr-Commit-Position: refs/heads/master@{#16286}
> Committed: 8e775e16ebTBR=henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7056
Review-Url: https://codereview.webrtc.org/2656533005
Cr-Commit-Position: refs/heads/master@{#16289}
WebRTC standalone build may depend on "video_capture_internal_impl"
instead of "video_capture". Including "video_capture" in public_deps
leads to duplicated definition in this case.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2657783002
Cr-Commit-Position: refs/heads/master@{#16279}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
This CL introduces a dedicated unit test for webrtc::RtpStreamReceiver.
Focus of this CL is testing RtpStreamReciver::OnReceivedPayloadData().
Dependencies with virtual interfaces are (g)mocked, non-virtual
dependencies are instantiated.
This CL is chained to https://codereview.webrtc.org/2638933002/ .
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2641463002
Cr-Commit-Position: refs/heads/master@{#16240}
It was only assigned at construction, and this improves consistency
with remote_estimator_proxy_.
The declaration of the private WrappingBitrateEstimator had to be
moved to the header file, and it was also converted from
system_wrappers' CriticalSectionWrapper to rtc::CriticalSection.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2642363003
Cr-Commit-Position: refs/heads/master@{#16236}
I've tested both the old and the new delay estimators and they all work ok at 15 kbps. The new estimators are a bit slower to adapt down so the maximum delay is a bit higher, especially at lower bitrates.
None of the estimators work at 10 kbps, but that is likely because the bitrate controller is configured to never go below 10 kbps. This means that it's impossible to empty the queues after a capacity drop to 10kbps (regardless of what the estimators do), so the delay stays high until the capacity increases.
BUG=webrtc:7022
Review-Url: https://codereview.webrtc.org/2644463009
Cr-Commit-Position: refs/heads/master@{#16233}
Reason for revert:
Triggers leak on Linux memcheck (non-default trybot):
### BEGIN MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)
Command: ../Release/./modules_unittests --isolated-script-test-output=/b/s/w/ioUlJCnu/output.json --isolated-script-test-chartjson-output=/b/s/w/ioUlJCnu/chartjson-output.json --gtest_filter=-CommonFormats/AudioProcessingTest*
Leak_DefinitelyLost
45 bytes in 1 blocks are definitely lost in loss record 118 of 277
operator new[](unsigned long) (m_replacemalloc/vg_replace_malloc.c:363)
webrtc::video_coding::H264SpsPpsTracker::CopyAndFixBitstream(webrtc::VCMPacket*) (/b/s/w/irJgAGsR/out/Release/modules_unittests)
webrtc::video_coding::TestH264SpsPpsTracker_SpsPpsOutOfBand_Test::TestBody() (/b/s/w/irJgAGsR/out/Release/modules_unittests)
Suppression (error hash=#0112A395AF2326BC#):
For more info on using suppressions see http://dev.chromium.org/developers/tree-sheriffs/sheriff-details-chromium/memory-sheriff#TOC-Suppressing-memory-reports
{
<insert_a_suppression_name_here>
Memcheck:Leak
fun:_Zna*
fun:_ZN6webrtc12video_coding17H264SpsPpsTracker19CopyAndFixBitstreamEPNS_9VCMPacketE
fun:_ZN6webrtc12video_coding42TestH264SpsPpsTracker_SpsPpsOutOfBand_Test8TestBodyEv
}
### END MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)
Original issue's description:
> H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
>
> - Changed method name to clarify that entire Nalus are expected.
> - Added unit test code.
> - Adjusted InsetSpsPpsNalus() implementation to above requirement.
>
> BUG=webrtc:5948
>
> Review-Url: https://codereview.webrtc.org/2638933002
> Cr-Commit-Position: refs/heads/master@{#16221}
> Committed: f53d7374cfTBR=philipel@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2649113003
Cr-Commit-Position: refs/heads/master@{#16225}
- Changed method name to clarify that entire Nalus are expected.
- Added unit test code.
- Adjusted InsetSpsPpsNalus() implementation to above requirement.
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2638933002
Cr-Commit-Position: refs/heads/master@{#16221}
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.
Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
> needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:
ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
In order to implicit cast an lvalue to an rvalue when returning
from a function, the return type and type of variable in the return
statement previously had to be exactly the same. When this was not
the case, std::move was required. For instance, when returning a
std::unique_ptr<Derived> variable in a function with a
std::unique_ptr<Base> return type, std::move is required.
DR 1579 changed this, and allows for implicitly converting
to the return type, if the return type has a constructor(T&&), where
T is the type of the local variable being returned. DR 1579 was
implemented in GCC 5, but not in GCC 4.9 and below. By explicitly
qualifying the local variable with std::move, we allow for compiling
with GCC 4.9 and incur no performance penalty. The code is still
absolutely correct to the word of C++11.
BUG=chromium:682965
See also:
* https://bugs.gentoo.org/show_bug.cgi?id=600288
* https://stackoverflow.com/questions/22018115/converting-stdunique-ptrderived-to-stdunique-ptrbase#comment33375875_22018521
* http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2014/n3833.html#1579
Review-Url: https://codereview.webrtc.org/2642053003
Cr-Commit-Position: refs/heads/master@{#16175}
Reason for revert:
Breaks tests downstream.
Original issue's description:
> Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
>
> Reason for revert:
> Fix in this CL: https://codereview.chromium.org/2640793003/
>
> Original issue's description:
> > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> >
> > Reason for revert:
> > Breaks android bots.
> >
> > Original issue's description:
> > > Make the new jitter buffer the default jitter buffer.
> > >
> > > This CL contains only the changes necessary to make the switch to the new jitter
> > > buffer, clean up will be done in follow up CLs.
> > >
> > > In this CL:
> > > - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > new video jitter buffer the default one.
> > > - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > >
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2627463004
> > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > Committed: 0f0763d86d
> >
> > TBR=stefan@webrtc.org,terelius@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/2632123005
> > Cr-Commit-Position: refs/heads/master@{#16117}
> > Committed: c08c191f7d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2642753002
> Cr-Commit-Position: refs/heads/master@{#16149}
> Committed: f20dd0014dTBR=stefan@webrtc.org,terelius@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/2638423003
Cr-Commit-Position: refs/heads/master@{#16159}
Reason for revert:
Fix in this CL: https://codereview.chromium.org/2640793003/
Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
>
> Reason for revert:
> Breaks android bots.
>
> Original issue's description:
> > Make the new jitter buffer the default jitter buffer.
> >
> > This CL contains only the changes necessary to make the switch to the new jitter
> > buffer, clean up will be done in follow up CLs.
> >
> > In this CL:
> > - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > new video jitter buffer the default one.
> > - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> >
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2627463004
> > Cr-Commit-Position: refs/heads/master@{#16114}
> > Committed: 0f0763d86d
>
> TBR=stefan@webrtc.org,terelius@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/2632123005
> Cr-Commit-Position: refs/heads/master@{#16117}
> Committed: c08c191f7dTBR=stefan@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2642753002
Cr-Commit-Position: refs/heads/master@{#16149}
Even though this is against the spec we allow a stream to continue if
a backwards jump in the picture id occurs on a keyframe.
BUG=webrtc:7001, webrtc:5514
Review-Url: https://codereview.webrtc.org/2640793003
Cr-Commit-Position: refs/heads/master@{#16146}
It combines and simplify use of GetStatusVector and GetReceiveDeltas accesors.
Replace use of all GetStatusVector/GetReceiveDeltasUs
BUG=None
Review-Url: https://codereview.webrtc.org/2633923003
Cr-Commit-Position: refs/heads/master@{#16139}
