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}
We have to move the content of the 'webrtc/build/' directory up of one level (see crbug.com/611808 for further details).
To avoid a collision with the DEPSed build/ at the top level we are moving all the content of 'webrtc/build/' to different locations (see webrtc:7030 for the suggested locations).
BUG=webrtc:7030
NOTRY=True
Review-Url: https://codereview.webrtc.org/2651913002
Cr-Commit-Position: refs/heads/master@{#16259}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
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}
They were passed on via RtpRtcp::Configuration, but unused for a
receive only RtpRtcp module.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2639423007
Cr-Commit-Position: refs/heads/master@{#16234}
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}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
- 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}
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.
webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.
Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
First step in moving the content of webrtc/build up one level.
BUG=webrtc:7030
Review-Url: https://codereview.webrtc.org/2653593003
Cr-Commit-Position: refs/heads/master@{#16219}
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}
https://codereview.webrtc.org/2514883002/ changed and moved these targets around but did not add public dependencies for the fallbacks, which causes gn gen --check a lot of anger.
NOTRY=true # Only build changes and windows bots are cranky atm.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2651663002
Cr-Commit-Position: refs/heads/master@{#16214}
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}
This script is similar to header_usage.sh, but it counts usage (number
of files) of each class defined in some header file.
E.g., the following classes appear unused,
AsyncHttpsProxyServerSocket
CompositeMediaEngineWithFakeVoiceEngine
DesktopId
RtpDataCallback
ScopedMessageData
ScreencastEventCatcher
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2649103002
Cr-Commit-Position: refs/heads/master@{#16204}
