In the metrics.h documentation the target to include a default
implementation of metrics was referring to the previous build system
(gyp). Now it is updated to refer to the current target.
BUG=None
NOTRY=True
TBR=henrika@webrtc.org
Review-Url: https://codereview.webrtc.org/2699093002
Cr-Commit-Position: refs/heads/master@{#16766}
Update the year in copyright headers from 2016 to 2017, and also rename a
variable in FallbackDesktopCapturerWrapperTest to follow coding style.
BUG=webrtc:7205
Review-Url: https://codereview.webrtc.org/2706193005
Cr-Commit-Position: refs/heads/master@{#16759}
plot_webrtc_test_logs.py: Add number of used cores to figure title.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2706753005
Cr-Commit-Position: refs/heads/master@{#16756}
AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(rate_calculator, bool limiter)
was added to create a mixer without the limiter subcomponent. Calling
it "Create with ... *and* limiter" is counterintuitive.
Renamed to simply 'Create'.
TBR=solenberg@webrtc.org
BUG=webrtc:7167
Review-Url: https://codereview.webrtc.org/2709523006
Cr-Commit-Position: refs/heads/master@{#16755}
This cl protects the access to the max_packet_size_, without fixing
the underlying race; the value is simply copied to a local variable,
whose value might be stale when used.
BUG=webrtc:7189
Review-Url: https://codereview.webrtc.org/2704263003
Cr-Commit-Position: refs/heads/master@{#16754}
Reason for revert:
Downstream fixed
Original issue's description:
> Revert of Fixes a bug where a video stream can get stuck in the suspended state. (patchset #8 id:120001 of https://codereview.webrtc.org/2705603002/ )
>
> Reason for revert:
> Breaks downstream
>
> Original issue's description:
> > Fixes a bug where a video stream can get stuck in the suspended state.
> >
> > This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough.
> >
> > This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before.
> >
> > BUG=webrtc:7178
> >
> > Review-Url: https://codereview.webrtc.org/2705603002
> > Cr-Commit-Position: refs/heads/master@{#16739}
> > Committed: a518a39963
>
> TBR=mflodman@webrtc.org,terelius@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:7178
>
> Review-Url: https://codereview.webrtc.org/2703393002
> Cr-Commit-Position: refs/heads/master@{#16751}
> Committed: b80bdcafedTBR=mflodman@webrtc.org,terelius@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:7178
Review-Url: https://codereview.webrtc.org/2704323003
Cr-Commit-Position: refs/heads/master@{#16753}
Reason for revert:
Breaks downstream
Original issue's description:
> Fixes a bug where a video stream can get stuck in the suspended state.
>
> This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough.
>
> This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before.
>
> BUG=webrtc:7178
>
> Review-Url: https://codereview.webrtc.org/2705603002
> Cr-Commit-Position: refs/heads/master@{#16739}
> Committed: a518a39963TBR=mflodman@webrtc.org,terelius@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:7178
Review-Url: https://codereview.webrtc.org/2703393002
Cr-Commit-Position: refs/heads/master@{#16751}
It's annoying to have to re-implement this every time I need a debug
printout.
Declared inline, so that there'll be zero runtime overhead.
This CL also modifies a unit test so that it will make use of the new
operator<< in case it finds errors.
BUG=none
Review-Url: https://codereview.webrtc.org/2705203002
Cr-Commit-Position: refs/heads/master@{#16749}
by creating it on accepted tmmbr/tmmbn rtcp messages
rather on sender/receiver reports.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2702373002
Cr-Commit-Position: refs/heads/master@{#16748}
Adds a public init function in RTCVideoFrame that makes it possible to
create a frame from a CVPixelBufferRef.
BUG=webrtc:7177
NOTRY=True
Review-Url: https://codereview.webrtc.org/2700113003
Cr-Commit-Position: refs/heads/master@{#16746}
Reason for revert:
Breaks downstream project.
Original issue's description:
> Add optional visualization file writers to VideoProcessor tests.
>
> The purpose of this visualization CL is to add the ability to record
> video at the source, after encode, and after decode, in the VideoProcessor
> tests. These output files can then be replayed and used as a subjective
> complement to the objective metric plots given by the existing Python
> plotting script.
>
> BUG=webrtc:6634
>
> Review-Url: https://codereview.webrtc.org/2700493006
> Cr-Commit-Position: refs/heads/master@{#16738}
> Committed: 872104ac41TBR=asapersson@webrtc.org,sprang@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2708103002
Cr-Commit-Position: refs/heads/master@{#16745}
The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping).
To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner.
The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter
Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other.
This also fixes a few minor GN issues so that warnings do not have to be suppressed.
BUG=webrtc:7167
Review-Url: https://codereview.webrtc.org/2692333002
Cr-Commit-Position: refs/heads/master@{#16742}
RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame, but it
currently contains some extra logic beyond that. We want RTCVideoFrame
to be as simple as possible, i.e. just a container with no extra state,
so we can use it as input to RTCVideoSource without complicating the
interface for consumers.
BUG=webrtc:7177
NOTRY=True
Review-Url: https://codereview.webrtc.org/2695203004
Cr-Commit-Position: refs/heads/master@{#16740}
This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough.
This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before.
BUG=webrtc:7178
Review-Url: https://codereview.webrtc.org/2705603002
Cr-Commit-Position: refs/heads/master@{#16739}
The purpose of this visualization CL is to add the ability to record
video at the source, after encode, and after decode, in the VideoProcessor
tests. These output files can then be replayed and used as a subjective
complement to the objective metric plots given by the existing Python
plotting script.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2700493006
Cr-Commit-Position: refs/heads/master@{#16738}
The video_capture module includes remnants of support for cameras
producing encoded frames. However, this seems to be unused, and is
explicitly not supported by VideoCaptureImpl::IncomingFrame.
BUG=None
Review-Url: https://codereview.webrtc.org/2668693008
Cr-Commit-Position: refs/heads/master@{#16732}
These tests involve interactions with the file system, so to avoid
flakiness they shouldn't be run in parallel.
BUG=webrtc:7195
NOTRY=True
Review-Url: https://codereview.webrtc.org/2710433003
Cr-Commit-Position: refs/heads/master@{#16727}
The partial availability problem aries from the fact that the minimum
supported OSX version is set to 10.9, but AppRTCMobile is using
functions available only in 10.10 and later. The minimum OSX version is
set as a declare_args() in build/config/mac/mac_sdk.gni, which makes it
difficult to override for just the AppRTCMobile target in WebRTC.
Instead, this CL solves the problem for now by removing the usage of the
10.10 function, which is trivial.
Also, the flag:
'extra_substitutions = [ "MACOSX_DEPLOYMENT_TARGET=10.8" ]'
is removed since it has no effect.
BUG=webrtc:4695
Review-Url: https://codereview.webrtc.org/2710493002
Cr-Commit-Position: refs/heads/master@{#16726}
Moves CameraCapturer, CameraSession, Camera1Session and Camera2Session
away from the public API.
BUG=webrtc:7172
Review-Url: https://codereview.webrtc.org/2699713004
Cr-Commit-Position: refs/heads/master@{#16723}
together with related functions and variables
to stress it is used for Tmmbr only.
This is explicitly pure rename CL with no functional changes.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2707763004
Cr-Commit-Position: refs/heads/master@{#16720}
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).
BUG=webrtc:6423
Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
