Reason for revert:
Not root cause for perf regression (regression still ongoing).
Original issue's description:
> Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
>
> Reason for revert:
> Speculative revert for perf regression.
>
> Original issue's description:
> > Remove VCMQmRobustness.
> >
> > Class contained a lot of not-really-wired-up functionality that ended up
> > being complicated ways of saying return 1; or return false;. This
> > removes this dependency that complicates code readability significantly.
> >
> > BUG=webrtc:5066
> > R=marpan@google.com, marpan@webrtc.org
> > TBR=stefan@webrtc.org
> >
> > Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> > Cr-Commit-Position: refs/heads/master@{#12516}
>
> TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5066, chromium:607838
>
> Committed: https://crrev.com/602316c3cd8556cc78d44f3ea4cd5fc8e70d9417
> Cr-Commit-Position: refs/heads/master@{#12572}
TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1941643002
Cr-Commit-Position: refs/heads/master@{#12583}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1935893002
Cr-Commit-Position: refs/heads/master@{#12577}
QuicDataChannel implements DataChannelInterface. It
replaces SCTP data channels by using a QuicTransportChannel
to create a ReliableQuicStream for each message.
QuicDataChannel only implements unordered, reliable delivery
for the initial implementation and does not send a hello message.
QuicDataTransport is a helper class that dispatches each incoming
ReliableQuicStream to a QuicDataChannel when the remote
peer receives a message by parsing the data channel id and message id
from the message header. It is also responsible for encoding the header
before QuicDataChannel sends the message.
Split from CL https://codereview.chromium.org/1844803002/.
BUG=
Review-Url: https://codereview.webrtc.org/1886623002
Cr-Commit-Position: refs/heads/master@{#12574}
The QuicWriteBlockedList needs to register outgoing QUIC
streams so that when the QuicTransportChannel becomes
unwritable and QUIC streams have buffered data, they can
send data once the QuicTransportChannel becomes writable.
Otherwise the QUIC streams will remain write blocked
after the QuicTransportChannel is writable.
BUG=
Review-Url: https://codereview.webrtc.org/1888903002
Cr-Commit-Position: refs/heads/master@{#12573}
Reason for revert:
Speculative revert for perf regression.
Original issue's description:
> Remove VCMQmRobustness.
>
> Class contained a lot of not-really-wired-up functionality that ended up
> being complicated ways of saying return 1; or return false;. This
> removes this dependency that complicates code readability significantly.
>
> BUG=webrtc:5066
> R=marpan@google.com, marpan@webrtc.org
> TBR=stefan@webrtc.org
>
> Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> Cr-Commit-Position: refs/heads/master@{#12516}
TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1935753002
Cr-Commit-Position: refs/heads/master@{#12572}
These changes are necessary to incorporate the latest
changes to QUIC sessions and the QUIC crypto handshake.
BUG=
Review-Url: https://codereview.webrtc.org/1910633003
Cr-Commit-Position: refs/heads/master@{#12571}
Objective C applications can now create new RTCRtpSenders and change their tracks, which gives them more fine grained control than MediaStreams.
BUG=
Review-Url: https://codereview.webrtc.org/1888633002
Cr-Commit-Position: refs/heads/master@{#12570}
NetEq bitexactness test depended on reference files which differs from platform to platform. This makes it very hard to update Neteq.
New method maintains the ability to save output into files. But it verifies the checksum only. With this, when bitexactness test fails, we can still check closely to the output file if need, but the test becomes much easier to modify.
BUG=
Review-Url: https://codereview.webrtc.org/1928923002
Cr-Commit-Position: refs/heads/master@{#12567}
And redefine rtc::Buffer as
using Buffer = BufferT<uint8_t>;
(In the long run, I'd like to remove the type alias and rename the
template to just rtc::Buffer, but that requires all current users of
Buffer to start saying Buffer<uint8_t> instead, and since Buffer is
used in the API, we can't do that in one step.)
The immediate reason for the new template is that we'd like to use
BufferT<int16_t> in the AudioDecoder interface.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/1929903002
Cr-Commit-Position: refs/heads/master@{#12564}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
Reason for revert:
Reverting this temporarily while I figure out the issues with the Chrome on android GN debug build.
Original issue's description:
> New task queueing primitive for async tasks: TaskQueue.
> TaskQueue is a new way to asynchronously execute tasks sequentially
> in a thread safe manner with minimal locking. The implementation
> uses OS supported APIs to do this that are compatible with async IO
> notifications from things like sockets and files.
>
> This class is a part of rtc_base_approved, so can be used by both
> the webrtc and libjingle parts of the WebRTC library. Moving forward,
> we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
>
> NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
> run on the same thread. E.g. on Mac and iOS, we use GCD dispatch queues
> which means that tasks might execute on different threads depending on
> what's the most efficient thing to do.
TBR=perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1935483002
Cr-Commit-Position: refs/heads/master@{#12562}
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking. The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.
This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library. Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread. E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.
Review-Url: https://codereview.webrtc.org/1919733002
Cr-Commit-Position: refs/heads/master@{#12561}
Reason for revert:
Chrome has been updated, cl https://codereview.chromium.org/1919283005/
Original issue's description:
> Revert of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #5 id:80001 of https://codereview.webrtc.org/1901973002/ )
>
> Reason for revert:
> GetYPlane, GetYPitch etc is used by Chromium.
>
> Original issue's description:
> > Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
> >
> > (And similarly for U and V). Also change video_frame_buffer method to
> > return a const ref to a scoped_ref_ptr.
> >
> > This cl is analogous to https://codereview.webrtc.org/1900673002/,
> > which delete corresponding methods in webrtc::VideoFrame.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/1c27c6bf4cf0476dd2f09425509afaae4cdfe599
> > Cr-Commit-Position: refs/heads/master@{#12492}
>
> TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/b05f994bb6f3055c852891c8acb531aee916a668
> Cr-Commit-Position: refs/heads/master@{#12494}
TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1923903002
Cr-Commit-Position: refs/heads/master@{#12559}
Reason for revert:
Breaks chrome FYI bots.
Original issue's description:
> Delete webrtc::VideoFrame methods buffer and stride.
>
> To make the HasOneRef/IsMutable hack work, also had to change the
> video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> to not imply an AddRef.
>
> BUG=webrtc:5682
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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:5682
Review-Url: https://codereview.webrtc.org/1935443002
Cr-Commit-Position: refs/heads/master@{#12558}
To make the HasOneRef/IsMutable hack work, also had to change the
video_frame_buffer method to return a const ref to a scoped_ref_ptr,
to not imply an AddRef.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1900673002
Cr-Commit-Position: refs/heads/master@{#12557}
VoENetwork is kept for now, but is not really used anylonger.
webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.
BUG=webrtc:5079
TBR=tommi
Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
This change replaces packet_iat_count_ms_ and max_timer_ms_, two
time-counting member variables in DelayManager, with Stopwatch objects
obtained from a TickTimer.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1929863002
Cr-Commit-Position: refs/heads/master@{#12554}
This CL will be followed up with another CL removing everything related
to RTP from ViEChannel to RtpStreamReceiver, i.e. remove ViEChannel::rtp_stream_receiver_.
BUG=5838
Review-Url: https://codereview.webrtc.org/1917363005
Cr-Commit-Position: refs/heads/master@{#12553}
Although the algorithm should generate the optimal gains, these limits are in place so that it doesn't go bananas in extreme noise situations or if there are some errors on the noise estimate. The limits were chosen as the extreme values from recordings with -3dB SNR, which is the minimum where the POLQA metric increases when processed.
Review-Url: https://codereview.webrtc.org/1925933002
Cr-Commit-Position: refs/heads/master@{#12550}
The render signal that reaches the IE is already normalized and compressed by an AGC on the far-end capture processing. Because the noise estimation is done before the AGC compression, we need to compensate for this gain in the IE to be able to compare these 2 powers.
Review-Url: https://codereview.webrtc.org/1913603002
Cr-Commit-Position: refs/heads/master@{#12549}
This is a first draft of what we're hoping to use to create all
AudioDecoder instances. Follow-up CLs will start using this internally
in NetEq instead of calling constructors manually.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/1917163002
Cr-Commit-Position: refs/heads/master@{#12548}
This CL adds the ability to convert RTCCertificate objects to and from
PEM string representations of it (its private key and certificate).
The RTCCertificate being a wrapper of SSLIdentity, this is where the
meat is.
Changes:
- SSLIdentity::PrivateKeyToPEMString() added. It together with the
already existing SSLCertificate::ToPEMString() yields both private
key and certificate PEM strings, both of which are required
parameters to SSLIdentity::FromPEMStrings().
- Its only implementation, OpenSSLIdentity::PrivateKeyToPemString().
- SSLIdentity::PublicKeyToPEMString() added, used by tests.
- sslidentity_unittest.cc updated:
* FromPEMStringsRSA and FromPEMStringsEC updated.
* CloneIdentityRSA and CloneIdentityECDSA added.
- RTCCertificate::To/FromPem added, using new class RTCCertificatePem.
- rtccertificate_unittest.cc: New test CloneWithPemSerialization.
- Renamed rtc_unittests.cc to rtccertificate_unittest.cc to match
convention.
BUG=webrtc:5794, chromium:581354
Review-Url: https://codereview.webrtc.org/1898383003
Cr-Commit-Position: refs/heads/master@{#12546}
