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:
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}
Reason for revert:
There seems an error made in this patch.
Hi Henrik,
I think the bit shift returned by CrossCorrelationWithAutoShift may be wrongly used by DotProduct.
We'd better revert this CL.
Doing another fix (and future fixes) will be paintful. I will work on a easy-to-modify bitexactness test first.
Original issue's description:
> Avoiding overflow in cross correlation in NetEq.
>
> BUG=
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/1925053002
Cr-Commit-Position: refs/heads/master@{#12543}
Specifically, this change replaces peak_period_counter_ms_ with
peak_period_stopwatch_ which obtains a Stopwatch object from
TickTimer. Necessary plumbing to get the TickTimer through to the
DelayPeakDetector is also included.
BUG=webrtc:5608
NOTRY=True
Review-Url: https://codereview.webrtc.org/1921163003
Cr-Commit-Position: refs/heads/master@{#12542}
Moves logic of default SocketServer from MessageQueue to SocketServer
Review-Url: https://codereview.webrtc.org/1891293002
Cr-Commit-Position: refs/heads/master@{#12541}
gn builds are missing Windows manifests for some webrtc executables.
These manifests are necessary in order to get predictable behavior on
Windows.
BUG=602505
NOTRY=True
Review-Url: https://codereview.webrtc.org/1927793002
Cr-Commit-Position: refs/heads/master@{#12540}
The voice engine expects send bitrates no more than the maximum for the
codec. For example, 510kbps for opus. So if "b=AS" sets a maximum above
the codec maximum, WebRtcVoiceEngine needs to cap it.
BUG=603690
Review-Url: https://codereview.webrtc.org/1920123002
Cr-Commit-Position: refs/heads/master@{#12537}
Changed DecoderDatabase::IsComfortNoise to do one, rather than four,
lookups of the rtp payload type. IsComfortNoise is called more frequently
since CNG was changed to not be an AudioDecoder.
BUG=606765
Review URL: https://codereview.webrtc.org/1923763003
Cr-Commit-Position: refs/heads/master@{#12533}
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/1923163003
Cr-Commit-Position: refs/heads/master@{#12532}
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/1930463002
Cr-Commit-Position: refs/heads/master@{#12530}
Reason for revert:
Breaks bots
Original issue's description:
> Deliver reordered packets in arrival time order after parsing feedback message.
>
> This is what the remote bitrate estimator expects.
>
> BUG=5823
> R=sprang@webrtc.org
>
> Committed: https://crrev.com/ed9535e2cac6be7a45fb03dc8426ef3b0ec43294
> Cr-Commit-Position: refs/heads/master@{#12527}
TBR=sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=5823
Review URL: https://codereview.webrtc.org/1928443004
Cr-Commit-Position: refs/heads/master@{#12528}
- Places most ObjC code into webrtc/sdk/objc instead.
- New gyp targets to build, strip and export symbols for dylib.
- Removes old script used to generate dylib.
BUG=
Review URL: https://codereview.webrtc.org/1903663002
Cr-Commit-Position: refs/heads/master@{#12524}
This fixes a problem where "b=AS" and "x-google-start-bitrate" can't
be used together. It also starts taking the minimum of "b=AS" and
"x-google-max-bitrate", instead of just letting "b=AS" win.
BUG=webrtc:5811
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1904063003 .
Cr-Commit-Position: refs/heads/master@{#12519}
Removes data race inside QualityScaler which were called both from the
codec thread and the invoking thread at the same time.
BUG=webrtc:5678
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1919883002 .
Cr-Commit-Position: refs/heads/master@{#12517}
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.orgTBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1917083003 .
Cr-Commit-Position: refs/heads/master@{#12516}
With this change, the NetEqImpl constructor takes a struct
(NetEqImpl::Dependencies) as input instead of a collection of
individual dependencies. The NetEqImpl unit test fixture is modified
to make better used of unique_ptrs.
Review URL: https://codereview.webrtc.org/1921243002
Cr-Commit-Position: refs/heads/master@{#12514}
