VideoTrackSource will be implemented in an upcoming cl but is needed to be included in libjingle.gyp in Chrome before the cl can be landed.
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1769343003 .
Cr-Commit-Position: refs/heads/master@{#11897}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
Changing from:
virtual void RequestIdentity(
rtc::KeyParams key_params,
rtc::Optional<uint64_t> expires,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
to:
virtual void RequestIdentity(
const rtc::KeyParams& key_params,
const rtc::Optional<uint64_t>& expires_ms,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
Making FakeDtlsIdentityStore DCHECK that |expires_ms| is not set, since it does not support that parameterization.
In a follow-up chromium CL the new signature will be used.
BUG=webrtc:5092, chromium:544902
Review URL: https://codereview.webrtc.org/1766673002
Cr-Commit-Position: refs/heads/master@{#11892}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
Sparse macro replaced for all audio histograms that have a constant name.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1762863003
Cr-Commit-Position: refs/heads/master@{#11885}
1. Fix the case of key frame accumulation being incorrect due to the chunk
size being computed at the time of leak based on input frame rate. The issue
is that the count is computed based on key frame ratio and the actual chunk
size computed from current input frame rate. These can be wildly different
especially at the beginning of the stream (key frame ratio defaults based
on 30 fps) resulting in incorrect key frame accumulation causing large frame
drops when the input frame rate is low.
2. Add large delta frame compensation. The current code accounts for key frames
but not large delta frames. This is a common occurence in some application
(remote desktop as an example)
3. Fixes an issue identified by the unit tests. The accumulation of
key frames had an issue in the scenario of a high key frame ratio where
the full key frame was not being accounted for.
3. Removes fast mode and other methods that are mostly dead code.
4. Cleans up variable names as per chromium style.
Review URL: https://codereview.webrtc.org/1750493002
Cr-Commit-Position: refs/heads/master@{#11884}
(the ones that were recently moved from c)
There are many files changed but most changes just
consist of adding namespaces.
In aec_common.h an C++-specific #ifdef needed to be added as
that file is both included from C and C++. I could see no
way around that but please let me know if there is a better
way around that.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1766663002
Cr-Commit-Position: refs/heads/master@{#11883}
Things done/implemented in this CL:
- An interface that can send Nack (VCMNackSender).
- An interface that can request KeyFrames (VCMKeyFrameRequestSender).
- The nack module (NackModule).
- A set of convenience functions for modular numbers (mod_ops.h).
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1715673002
Cr-Commit-Position: refs/heads/master@{#11882}
This fixes a couple major issues.
#1: If the payload type that an RTX codec refers to has been reassigned, and then the RTX codec is added in a subsequent offer, it refers to the wrong payload type.
#2: If we receive an offer with two payload types referring to the same codec (which we support), our answer contains both (instead of just one), which causes issues down the road since the video engine only supports one payload type per codec.
BUG=webrtc:5450,webrtc:5499
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1616033002 .
Cr-Commit-Position: refs/heads/master@{#11880}
Soft reset can be used when input frame resolution changes
to avoid re creating MediaCodec instance.
Instead MediaCodec is flushed and some variables are reset.
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1732533002 .
Cr-Commit-Position: refs/heads/master@{#11878}
api function that directly returns aec_core.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1695743004
Cr-Commit-Position: refs/heads/master@{#11875}
With this change, NetEq now uses AudioFrame as output type, like the
surrounding functions in ACM and VoiceEngine already do.
The computational savings is probably slim, since one memcpy is
removed while another one is added (both in AcmReceiver::GetAudio).
More simplifications and clean-up will be done in
AcmReceiver::GetAudio in future CLs.
BUG=webrtc:5607
Review URL: https://codereview.webrtc.org/1750353002
Cr-Commit-Position: refs/heads/master@{#11874}
This new class allows usage of a QuicSession to establish a QUIC handshake.
BUG=
Review URL: https://codereview.webrtc.org/1721673004
Cr-Commit-Position: refs/heads/master@{#11873}
This temporarily fixes a probem where renderers causes VideoSinkWants.rotation_applied=true.
The problem was introduced by https://codereview.webrtc.org/1759473003/ where VideTrackRenderes are registered to the cricket::VideoCapturer with default VideoSinkWants.
BUG=webrtc:5621
Review URL: https://codereview.webrtc.org/1764693004
Cr-Commit-Position: refs/heads/master@{#11871}
all interfaces that formerly took constraints parameters
in name=value form.
This is in preparation for making Chrome only use these
explicit interfaces.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1717583002
Cr-Commit-Position: refs/heads/master@{#11870}
Renamed the new variant of EncodeInternal to EncodeImpl, so that
subclasses implementing one of the EncodeInternal don't need to
explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their
implementation hiding the other variant of EncodeInternal. This causes
a warning (treated as an error) when building using GCC.
Review URL: https://codereview.webrtc.org/1764583003
Cr-Commit-Position: refs/heads/master@{#11868}
The steps involved were:
1) Change file name to .cc from .c.
2) Update the build files accordingly.
3) Remove the extern header file inclusion.
4) Change the casts in aec_resampler.cc to static_cast
and reinterpret_cast.
The changes are bitexact.
The CL will be followed with another CL where a proper (webrtc) namespace is introduced. The reason for not having it in this CL is that this was missed in the corresponding
CL that did the above for aec_core.c, ..., and if the
namespaces in all the aec_core -related files can be changed
at the same time that will simplify things.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1754223004
Cr-Commit-Position: refs/heads/master@{#11867}
audio processing module.
The test also adds a new helper class called
VectorBasedAudioFrame that is intended to be
reused for the bitexactness tests for the other
submodules.
BUG=webrtc:1091
Review URL: https://codereview.webrtc.org/1510493004
Cr-Commit-Position: refs/heads/master@{#11864}
that was not updated as it should.
The bug caused no negative impact at all apart
from a missed check that the Aec handles above
index 0 were not null. That check is, however,
done elsewhere so there was no negative impact
of this bug.
BUG=
Review URL: https://codereview.webrtc.org/1716203002
Cr-Commit-Position: refs/heads/master@{#11863}
This is useful to keep the NAT binding alive on backup connections.
BUG=
Review URL: https://codereview.webrtc.org/1737493004
Cr-Commit-Position: refs/heads/master@{#11862}
Makes VideoCaptureInput easier to test and enables running more things
outside VideoCaptureInput on the encoder thread in the future
(initializing encoders and reconfiguring them, for instance).
BUG=webrtc:5410, webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1763693002 .
Cr-Commit-Position: refs/heads/master@{#11860}
Intended to make SetEncoder callable from another thread so that
ReconfigureVideoEncoder can post SetEncoder over and return earlier to
prevent blocking the calling thread.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1751363003 .
Cr-Commit-Position: refs/heads/master@{#11856}
This CL replaces the function SurfaceTextureHelper.setListener() that
could only be called once with the functions startListening() and
stopListening() that can be called multiple times. This is necessary
when the SurfaceTextureHelper will be passed to the VideoCapturerAndroid
in startCapture(). startListening() will be called in startCapture() and
stopListening() in stopCapture().
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1755573002
Cr-Commit-Position: refs/heads/master@{#11855}
This produces bit-exact output and doesn't have the performance sensitivity to vectorisation, giving a complexity decrease of the IntelligibilityEnhancer of about 30x in my local machine.
This performance issue was put in evidence by this CL: https://codereview.webrtc.org/1693823004/
BUG=590998
Review URL: https://codereview.webrtc.org/1755943002
Cr-Commit-Position: refs/heads/master@{#11851}
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.
BUG=webrtc:4591
Review URL: https://codereview.webrtc.org/1577233006
Cr-Commit-Position: refs/heads/master@{#11850}
Move the "webrtc_test_common" target to test.gyp and rename
it to "test_common".
Move all tests in "webrtc_test_common_unittests" (which
wasn't run on the bots) into "test_support_unittests".
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1754593002
Cr-Commit-Position: refs/heads/master@{#11848}
Also moves and simplifies SetSendCodec from VideoSendStream to mostly
inside ViEEncoder. This is necessary for making
ReconfigureVideoEncoder asynchronous as we don't post any result back.
BUG=webrtc:5494
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1754283002 .
Cr-Commit-Position: refs/heads/master@{#11847}
