This layer takes in a simplified "options" struct and the current local description,
and generates a new offer/answer. Previously the options struct assumed there would
only be one media description per media type (audio/video), but it now supports
N number of audio/video descriptions.
The |add_legacy_stream| options is removed from the mediasession.cc/.h
in this CL.
The next step is to add the ability for PeerConnection/WebRtcSession to create
"options" to represent multiple RtpTransceivers, and apply the Unified Plan
descriptions correctly. Right now, only Plan B descriptions will be
generated in unit tests.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Commit-Position: refs/heads/master@{#19343}
This CL adds the ability for a SSLAdapter to resume a previous session, saving a roundtrip and significantly reducing the # of bytes needed to bring up the new session.
To do this, the sessions need to share state. This is addressed by introducing the SSLAdapterFactory object, which can maintain a SSL_CTX and session cache for multiple sessions.
This CL does not have unit tests in order to minimize the change size (i.e., to reduce the size of the CP). CL https://chromium-review.googlesource.com/c/558612 builds on this CL and adds tests, but makes some nontrivial changes to SSLStreamAdapter in order to get the test server to share a SSL_CTX across sessions.
Bug: 7936
Change-Id: I677b73453d981d5b3a2e66ea9a5be722acd59475
Reviewed-on: https://chromium-review.googlesource.com/575910
Commit-Queue: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19342}
Given the current state of OpenSLES (disabled in many places), making
this a debug line makes more sense than an error.
BUG=none
Change-Id: I16d46d3f8234ebeffe820d92e7a6d7ed3eae11cd
Reviewed-on: https://chromium-review.googlesource.com/611491
Commit-Queue: Henrik Andreasson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19340}
Reason for revert:
Speculative revert to see if this caused regressions in android perf tests.
Original issue's description:
> Add functionality which limits the number of bytes on the network.
>
> The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
>
> Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
>
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/2918323002
> Cr-Commit-Position: refs/heads/master@{#19289}
> Committed: 8497fdde43TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/3001653002
Cr-Commit-Position: refs/heads/master@{#19339}
Rvalue reference arguments are generally banned by the style guide.
Bug: None
Change-Id: I4314859623ffcf056f53c42087b59696b5e71690
Reviewed-on: https://chromium-review.googlesource.com/531028
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Michael T <tschumim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19338}
Reason for revert:
Speculative revert to see if this caused regressions in android perf tests.
Original issue's description:
> Make the acceptable queue in the cwnd experiment configurable.
>
> BUG=webrtc:7926
>
> Review-Url: https://codereview.webrtc.org/2998753002
> Cr-Commit-Position: refs/heads/master@{#19320}
> Committed: 7c83c56b6dTBR=philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2999893002
Cr-Commit-Position: refs/heads/master@{#19337}
VideoSinks receive the new kind of VideoFrames and will replace
VideoRenderers. Converting from old texture frames to VideoFrames will
involve conversion to I420 so it is not recommended to use VideoSinks
before all sources produce VideoFrames.
BUG=webrtc:7749, webrtc:7760
Review-Url: https://codereview.webrtc.org/3002553002
Cr-Commit-Position: refs/heads/master@{#19335}
This CL completely removes the methods
AudioProcessing::{Start,Stop}DebugDumpRecording. These methods have
been replaced with AudioProcessing::{Attach,Detach}AecDump. Their
implementation was removed in the parent CL
https://chromium-review.googlesource.com/c/589147
Bug: webrtc:7404
Change-Id: Ia3d5314985af9c74f79c94c514ded1f8afc78fb5
Reviewed-on: https://chromium-review.googlesource.com/589152
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19334}
AudioProcessingModule has a feature to make a recording of its
configuration, inputs and outputs over a period of time. In the past
CLs, this feature has been rewritten to move file IO away from
real-time audio threads. The interface has changed from
{Start,Stop}DebugDumpRecording to {Attach,Detach}AecDump.
This CL removes the previous implementation of the old interface
StartDebugRecording. The public interface is left to not cause
problems to downstream projects. It will be removed in the dependent
CL https://chromium-review.googlesource.com/c/589152/
With this CL, usage of WEBRTC_AUDIOPROC_DEBUG_DUMP and ~300 LOC of
logging code is removed from AudioProcessingImpl.
Bug: webrtc:7404
Change-Id: I16e7b557774e4bc997e1f5de4f97ed2c31d63879
Reviewed-on: https://chromium-review.googlesource.com/589147
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19332}
Also remove |key_frame_interval| from argument list, since it is always
set to -1.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2999643002
Cr-Commit-Position: refs/heads/master@{#19331}
* Don't loop over fps, but do loop over codec implementation type.
* Order codec settings as they are defined in the test.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/3000613002
Cr-Commit-Position: refs/heads/master@{#19330}
This is never used in WebRTC, so we can probably remove it.
BUG=webrtc:8082
NOTRY=True
Review-Url: https://codereview.webrtc.org/2995673002
Cr-Commit-Position: refs/heads/master@{#19329}
Reason for revert:
Create fix CL.
Original issue's description:
> Revert of Request keyframes more frequently on stream start/decoding error. (patchset #1 id:1 of https://codereview.webrtc.org/2993793002/ )
>
> Reason for revert:
> Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
>
> Original issue's description:
> > Request keyframes more frequently on stream start/decoding error.
> >
> > In this CL:
> > - Added FrameObject::is_keyframe() convinience function.
> > - Moved logic to request keyframes on decoding error from VideoReceived to
> > VideoReceiveStream.
> > - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
> >
> > BUG=webrtc:8074
> >
> > Review-Url: https://codereview.webrtc.org/2993793002
> > Cr-Commit-Position: refs/heads/master@{#19280}
> > Committed: 26b4804358
>
> TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2994043002
> Cr-Commit-Position: refs/heads/master@{#19295}
> Committed: 77a983185fTBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2996823002
Cr-Commit-Position: refs/heads/master@{#19324}
fake_decoder (of type test::FakeDecoder) was not used.
BUG=None
Review-Url: https://codereview.webrtc.org/3001543002
Cr-Commit-Position: refs/heads/master@{#19322}
This flag enables support for Android Studio 3.0 which allows us to use
Java 8 features. Gradle is updated to version 4.1.0.
BUG=webrtc:8084
Review-Url: https://codereview.webrtc.org/2994123002
Cr-Commit-Position: refs/heads/master@{#19319}
This change implements GetWindowList() on X11. WindowCapturerLinux and
GetWindowUnderPoint() can share the logic of this function.
Bug: webrtc:7950
Change-Id: Ida746840d6f51d31e0470e5ae4955b6f5a4cfaf2
Reviewed-on: https://chromium-review.googlesource.com/606560
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#19314}
* EndToEndTest.InitialProbing had an uninitialized boolean.
* Both tests used RTC_DCHECK where one would normally expect an RTC_DCHECK.
BUG=webrtc:8085
Review-Url: https://codereview.webrtc.org/2998793002
Cr-Commit-Position: refs/heads/master@{#19309}
There exist a bug in the video_coding::PacketBuffer which triggers when a
frame is the same size as the buffer. A trivial workaround is to increase
the start size to something big so that this never happens in practice.
The bug has been fixed but we still want to test the workaround in ToT,
which is why this CL exist.
BUG=webrtc:8028, chromium:752886
R=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2994093002 .
Cr-Commit-Position: refs/heads/master@{#19308}
This CL brings us one step closer to removing CodecDatabase and
GenericEncoder, by removing the static VCM::Codec(). Codec specific
methods are moved to video_encoder.cc (they already belonged to this
class) and getting default generic codec settings has been moved to a
test specific file.
This CL also makes video_encoder.h pass style guide and lint checks,
since these checks are triggered with the new video_encoder.cc file.
BUG=webrtc:8064
Review-Url: https://codereview.webrtc.org/2993923002
Cr-Commit-Position: refs/heads/master@{#19303}
That way, the debug printout will tell us which of x and y that was false.
BUG=none
Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
Reason for revert:
Broke downstream test that was waiting for 5 keyframes to be received within 10 seconds. Maybe the issue is that "stats_callback_->OnCompleteFrame(frame->num_references == 0, ..." was changed to "frame->is_keyframe()"?
Original issue's description:
> Request keyframes more frequently on stream start/decoding error.
>
> In this CL:
> - Added FrameObject::is_keyframe() convinience function.
> - Moved logic to request keyframes on decoding error from VideoReceived to
> VideoReceiveStream.
> - Added keyframe_required as a parameter to FrameBuffer::NextFrame.
>
> BUG=webrtc:8074
>
> Review-Url: https://codereview.webrtc.org/2993793002
> Cr-Commit-Position: refs/heads/master@{#19280}
> Committed: 26b4804358TBR=terelius@webrtc.org,stefan@webrtc.org,noahric@chromium.org,philipel@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2994043002
Cr-Commit-Position: refs/heads/master@{#19295}
When compiling webrtc's call.cc with VC++ 2017 (is_clang = false) the
following compile error occurs:
sequence_number_util.h(90): error C2672: 'rtc::SafeLt': no matching
overloaded function found
note: see reference to class template instantiation
'webrtc::SeqNumUnwrapper<T,M>' being compiled
This error is not associated with any particular instantiation of
SeqNumUnwrapper (there isn't one) and this undefined nature of 'T' seems
to be what confuses the compiler. When it tries to locate SafeLt for an
undefined type 'T' it gets confused.
SafeLt is unnecessary in this context and changing it to use the '<'
operator directly avoids the problem.
The bug has been reported to Microsoft.
BUG=chromium:753488
Review-Url: https://codereview.webrtc.org/2997623002
Cr-Commit-Position: refs/heads/master@{#19292}
This has been a frequent source of confusion, especially since the
method names don't match anything in the standard exactly.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2994733002
Cr-Commit-Position: refs/heads/master@{#19290}
The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt.
Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds).
BUG=webrtc:7926
Review-Url: https://codereview.webrtc.org/2918323002
Cr-Commit-Position: refs/heads/master@{#19289}
[This CL is work in progress.]
Wire up the rtp keep-alive in webrtc::Call::Config using new
SetRtpTransportParameters() method on RtpTransportInterface.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2981513002
Cr-Commit-Position: refs/heads/master@{#19287}
In this CL:
- Added FrameObject::is_keyframe() convinience function.
- Moved logic to request keyframes on decoding error from VideoReceived to
VideoReceiveStream.
- Added keyframe_required as a parameter to FrameBuffer::NextFrame.
BUG=webrtc:8074
Review-Url: https://codereview.webrtc.org/2993793002
Cr-Commit-Position: refs/heads/master@{#19280}
The AsyncInvoker destructor waits for all invoked tasks to be complete
(in other words, all AsyncClosures to be destructed). They were using an
event to wake up the destructor, but a race made it possible for this
event to be dereferenced after it's destroyed.
This CL makes the event reference counted, such that if the destructor
runs right after AsyncClosure decrements "pending_invocations_",
setting the event will be a no-op, and the event will be destructed
in the AsyncClosure destructor.
This CL also fixes a deadlock that may occur for "re-entrant"
invocations. The deadlock occurs if the AsyncInvoker is destroyed on
thread A while a task on thread B is running, which AsyncInvokes a task
back on thread A.
This was causing pending_invocations_ to end up negative, because
an AsyncClosure that's never added to a thread's message queue (due to
the "destroying_" flag) caused the count to be decremented but not
incremented.
BUG=webrtc:7656
Review-Url: https://codereview.webrtc.org/2885143005
Cr-Commit-Position: refs/heads/master@{#19278}
The existing unity plugin (an example in webrtc codebase) does not support camera access on Android platform. This CL implements such functionality.
TBR=gyzhou@chromium.org
BUG=webrtc:8067
Review-Url: https://codereview.webrtc.org/2993273002
Cr-Commit-Position: refs/heads/master@{#19277}
During window capture on Windows 10, if the selected window is minimized,
ShouldUseScreenCapturer() still thinks it's on top and continue to do a
screencapture which is meaningless.
This cl will set |.is_top_window| with false to minimized window,then we
can skip doing any capture to it.
BUG=chromium:568835
Review-Url: https://codereview.webrtc.org/2997493002
Cr-Commit-Position: refs/heads/master@{#19276}
