Before only C420 as format name was accepted, now C420mpeg2 is also
accepted. Both means the same thing.
BUG=webrtc:6545
NOTRY=True
Review-Url: https://codereview.webrtc.org/2468943002
Cr-Commit-Position: refs/heads/master@{#14897}
In the new APM statistics interface, the default values did not match those previously used in AudioSendStream::Stats.
BUG=webrtc:6525
Review-Url: https://codereview.webrtc.org/2469783002
Cr-Commit-Position: refs/heads/master@{#14896}
This fix is made to remove the discrepancy between GYP and GN audio_decoder_factory_interface target.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2472643003
Cr-Commit-Position: refs/heads/master@{#14894}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Original-Commit-Position: refs/heads/master@{#14887}
Cr-Commit-Position: refs/heads/master@{#14892}
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"
The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.
Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.
- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).
BUG=webrtc:6579
Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
Reason for revert:
Landed the wrong patchset. Nothing broken.
Original issue's description:
> Remove webrtc::Video from H264 encoder internals
>
> This CL replaces the use of webrtc::Video as an internal
> variable in the H.264 encoder with the specific fields
> that are used by this encoder.
>
> In support of refactorings discussed around:
>
> BUG=600254
>
> Committed: https://crrev.com/2549437b5ccf5aae2e6f1a1491c5f505d1859f9c
> Cr-Commit-Position: refs/heads/master@{#14887}
TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254
Review-Url: https://codereview.webrtc.org/2472673002
Cr-Commit-Position: refs/heads/master@{#14888}
This CL replaces the use of webrtc::Video as an internal
variable in the H.264 encoder with the specific fields
that are used by this encoder.
In support of refactorings discussed around:
BUG=600254
Review-Url: https://codereview.webrtc.org/2468903003
Cr-Commit-Position: refs/heads/master@{#14887}
Reason for revert:
Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio
Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.
Original issue's description:
> Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
>
> Replaced with webrtc::VideoFrame.
>
> TBR=mflodman@webrtc.org
> BUG=webrtc:5682
>
> Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> Cr-Commit-Position: refs/heads/master@{#14885}
TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@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/2471783002
Cr-Commit-Position: refs/heads/master@{#14886}
Reason for revert:
Prevents WebRTC rolls into Chrome.
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%29/builds/14848/steps/compile/logs/stdio
The reason for reverting is: Breaks
https://build.chromium.org/p/chromium.linux/builders/Blimp%20Linux%20%28dbg%2...
[881/894] SOLINK ./libcontent.so
FAILED: libcontent.so libcontent.so.TOC
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:45: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawWindowCapturer(webrtc::DesktopCaptureOptions
const&)'
../../third_party/webrtc/modules/desktop_capture/desktop_capturer.cc:56: error:
undefined reference to
'webrtc::DesktopCapturer::CreateRawScreenCapturer(webrtc::DesktopCaptureOptions
const&)'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.
Original issue's description:
> Add CreateWindowCapturer() and CreateScreenCapturer() in DesktopCapturer
>
> This change copies ScreenCapturerDifferWrapper to a new
> DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
> DesktopCapturer::CreateScreenCapturer functions to replace
> WindowCapturer::Create and ScreenCapturer::Create.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/b763e39beba92b45baa09542f949daabbe6258a3
> Cr-Commit-Position: refs/heads/master@{#14880}
TBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2471773002
Cr-Commit-Position: refs/heads/master@{#14884}
They can be removed and we can use the default system controls.
It's less code and also has more native look.
BUG=webrtc:6617
Review-Url: https://codereview.webrtc.org/2455413002
Cr-Commit-Position: refs/heads/master@{#14882}
To achieve this, several changes needed to be made on both UI and
app logic level.
* Settings view controller is added (modally shown when the settings
button is pressed).
- From there the user can see the current capture resolution
and select another capture resolution.
* Model class for the capture resolution added.
- Improves readability and makes separation of concerns cleaner
- Handles persisting
- Provides defaults
- Maps video resolution setting to RTCMediaConstraints dictionary
* Test for the model class
In future it would be possible to extend this CL and add further settings (i.e
bit rate).
Also it would be easy to remove the hardcoded resolutions and use dynamic values
depending on device capability.
BUG=webrtc:6473
Review-Url: https://codereview.webrtc.org/2462623002
Cr-Commit-Position: refs/heads/master@{#14881}
This change copies ScreenCapturerDifferWrapper to a new
DesktopCapturerDifferWrapper, and adds DesktopCapturer::CreateWindowCapturer and
DesktopCapturer::CreateScreenCapturer functions to replace
WindowCapturer::Create and ScreenCapturer::Create.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Commit-Position: refs/heads/master@{#14880}
With this change, instead of
RTC_DCHECK_GE(unsigned_var, 17u);
we can simply write
RTC_DCHECK_GE(unsigned_var, 17);
or even
RTC_DCHECK_GE(unsigned_var, -17); // Always true.
and the mathematically sensible thing will happen.
Perhaps more importantly, we can replace checks like
// index is size_t, num_channels is int.
RTC_DCHECK(num_channels >= 0
&& index < static_cast<size_t>(num_channels));
or, even worse, just
// Surely num_channels isn't negative. That would be absurd!
RTC_DCHECK_LT(index, static_cast<size_t>(num_channels));
with simply
RTC_DCHECK_LT(index, num_channels);
In short, you no longer have to keep track of the signedness of the arguments, because the sensible thing will happen.
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2459793002
Cr-Commit-Position: refs/heads/master@{#14878}
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.
To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
In this CL:
- Don't insert a packet if we have explicitly cleared past it.
- Added some logging to ExpandBufferSize.
- Renamed IsContinuous to PotentialNewFrame.
- Unittests updated/added for this new behavior.
- Refactored TestPacketBuffer unittests.
BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2399373002 .
Cr-Commit-Position: refs/heads/master@{#14871}
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
Writable() and the related signal are already part of rtc::PacketTransportInterface. Sense of code symmetry aesthetics dictates that receiving() and the related signal should be declared in the same place.
BUG=webrtc:6531
Review-Url: https://codereview.webrtc.org/2444793003
Cr-Commit-Position: refs/heads/master@{#14865}
Reason for revert:
Breaks Chrome FYI.
peerconnection_unittest calls RTCStatsReport::Create without parameters.
Original issue's description:
> RTCOutboundRTPStreamStats[1] added.
>
> This also adds RTCRTPStreamStats[2] which it derives from. Not all stats
> are supported in this CL, this must be addressed before closing the
> issue.
>
> RTCStatsReport also gets a timestamp and ToString.
>
> [1] https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
> [2] https://w3c.github.io/webrtc-stats/#streamstats-dict*
>
> BUG=chromium:627816, chromium:657856, chromium:657854
>
> Committed: https://crrev.com/69e9cb08285f6cbcab547c7a5e6aa668fa6f2d29
> Cr-Commit-Position: refs/heads/master@{#14860}
TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:627816, chromium:657856, chromium:657854
Review-Url: https://codereview.webrtc.org/2465223002
Cr-Commit-Position: refs/heads/master@{#14863}
Reason for revert:
Dependend cl Breaks Chrome FYI.
peerconnection_unittest anropar RTCStatsReport::Create without parameters.
Original issue's description:
> RTCInboundRTPStreamStats[1] added.
>
> Not all stats are collected in this CL, this must be addressed before
> closing the issue.
>
> [1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
>
> BUG=chromium:627816, chromium:657855, chromium:657854
>
> Committed: https://crrev.com/0d7bf169402ea9345d163998f4f7df89229ac470
> Cr-Commit-Position: refs/heads/master@{#14861}
TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:627816, chromium:657855, chromium:657854
Review-Url: https://codereview.webrtc.org/2470683002
Cr-Commit-Position: refs/heads/master@{#14862}
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.
BUG=chromium:653268
Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
Contains fixes for a non-perfect implementation in https://codereview.webrtc.org/2328433003/
Summary:
Adds WebRTC.Audio.RecordedOnlyZeros UMA stat when recording stops if:
- All level estimates during the audio session were zero, and
- If the audio session was longer than 10 seconds.
Adds four simple methods to the AudioDeviceBuffer (ADB) class to allow the ADM
to update the ADB about when media starts and stops in both directions.
Moves any "critical" parst out frome the timer (based on task queue) and ensures
that it only does trivial logging tasks.
The task queue is now owned by a unique pointer to improve control of when it
starts and stops.
Adds time measurements (for logging) of both total time playing out and total
recording time. Units are in milliseconds.
BUG=webrtc:6592
Review-Url: https://codereview.webrtc.org/2445363003
Cr-Commit-Position: refs/heads/master@{#14854}
The new code is only exercised in tests so far. The H264 profile-level-id
parsing is not complete, but it should be enough for our purposes for
now.
BUG=webrtc:6400,webrtc:6337
Review-Url: https://codereview.webrtc.org/2459633002
Cr-Commit-Position: refs/heads/master@{#14850}
