This change replaces all GetWindowList / GetScreenList with GetScreenList,
SelectWindow / SelectScreen with SelectSource, and BringSelectedWindowToFront
with FocusOnSelectedSource in WebRTC.
BUG=webrtc:6513
Review-Url: https://codereview.webrtc.org/2479553006
Cr-Commit-Position: refs/heads/master@{#14960}
Half of the function's code was reformatted and reindented in
https://codereview.webrtc.org/1307663004, but the bottom half was still
adhering to an old coding style and using different indentation values.
Not only does this make the code look confusing, but it can cause build
issues on certain compilers: for example, GCC 6.2.0 with -Wall causes
the build to fail because -Wmisleading-indentation is enabled.
BUG=None
R=asapersson@webrtc.org,danilchap@webrtc.org
Review-Url: https://codereview.webrtc.org/2479193002
Cr-Commit-Position: refs/heads/master@{#14957}
This CL removes the coupling of the VideoToolbox h264 implementation
to the generic h264 code. The files have been moved into sdb/obj/Framework
and all dependency on them has been removed from the rest of WebRTC.
We now add it as an external encoder via a factory supplied to the
CreatePeerConnectionFactory call. This also brings the iOS implementation
closer to what we do on Android for MediaCodec.
BUG=webrtc:6619
Review-Url: https://codereview.webrtc.org/2463313002
Cr-Commit-Position: refs/heads/master@{#14953}
- Removes the lock that was used to protect the audio transport object.
It is now protected "by design" instead.
- Removes rec/play_bytes_per_sample_ since we only support 16-bit samples.
BUG=webrtc:6560
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/2466613002 .
Cr-Commit-Position: refs/heads/master@{#14950}
This change the interface of the PacketBuffer since the bitstream data of the packet has to be persistent when inserted into the PacketBuffer.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2476283002 .
Cr-Commit-Position: refs/heads/master@{#14949}
To differentiate between when a packet should be dropped or when a keyframe
is missing its SPS and/or SPS CopyAndFixBitstream now returns:
- kInsert, the packet should be inserted into the PacketBuffer.
- kDrop, the packet should be dropped.
- kRequestKeyframe, a keyframe should be requested.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2477343002
Cr-Commit-Position: refs/heads/master@{#14948}
- Change const ptr to const ref in parameter list.
Using nullptr as argument was invalid, so no need to send
pointer instead of reference.
- Change return type to void or bool, where appropriate
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2455963003
Cr-Commit-Position: refs/heads/master@{#14945}
Prior to this change, we signalled that ULPFEC was disabled
through a bool, but that RED was disabled by setting its
payload type to -1. The latter is consistent with how we
disable RED/ULPFEC in the config, so this CL removes the
ULPFEC bool from the {,Set}UlpfecConfig chain of member
functions.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2460533002
Cr-Commit-Position: refs/heads/master@{#14944}
At the same time, change to using int's instead of uint8_t's for the payload type.
This allows us to signal disabled FEC or RED using the sentinel value -1, which
is commonplace in other parts of the code.
These APIs will be deprecated when ULPFEC is deprecated.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2448463003
Cr-Commit-Position: refs/heads/master@{#14942}
Since ffmpeg use an optimized bitstream reader that reads in chunks of 32/64
bits the bitstream buffer has to be increased in order for the reader to not
read out of bounds.
BUG=webrtc:5514
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2476513004 .
Cr-Commit-Position: refs/heads/master@{#14941}
Also improve denoiser to not assume identical stride of all involved
frames, and delete the no longer needed function I420Buffer::CopyKeepStride.
BUG=None
Review-Url: https://codereview.webrtc.org/2469763002
Cr-Commit-Position: refs/heads/master@{#14940}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Original-Commit-Position: refs/heads/master@{#14923}
Cr-Commit-Position: refs/heads/master@{#14935}
Previously probing bitrate was capped at 10Mbps, which is too low for some
application. Now ProbeContoller limits max probing rate to max allowed
bitrate, which can be specified by the application.
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2430133005
Cr-Commit-Position: refs/heads/master@{#14927}
Reason for revert:
Still breaks internal downstream project.
Sergey: Please update internal project before relanding this.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/c681250aaa2025836db7669694e323898e5c2ca7
> Cr-Commit-Position: refs/heads/master@{#14923}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2479643002
Cr-Commit-Position: refs/heads/master@{#14925}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14923}
This class will interface RTPSenderVideo with the underlying
erasure code. It is functionally similar to ProducerFec
(to be renamed UlpfecGenerator). In fact, the FlexfecSender is a
friend of ProducerFec, and reuses most of its implementation.
Besides the fact that FlexfecSender outputs FlexFEC packets,
the main difference with ProducerFec is that FlexfecSender
allocates RTP sequence numbers, whereas ProducerFec does not
do this for the RED-encapsulated ULPFEC packets.
This class is split as interface/implementation, since it will
be owned by VideoSendStream initially. Further along, it may be
owned by PacedSender.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2441613002
Cr-Commit-Position: refs/heads/master@{#14922}
Since it is unsafe to hand out a pointer to a packet that might be removed/
overwritten at any time we now return a copy of the header if it exist.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2468183002
Cr-Commit-Position: refs/heads/master@{#14920}
There is no need for it to be an interface.
In this CL, I also took the opportunity to make two small fixes:
- remove the 'flexfec_' prefix from some member variables
- remove unnecessary use of a stringstream object
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2471073003
Cr-Commit-Position: refs/heads/master@{#14919}
Reason for revert:
Breaks everything
Original issue's description:
> Revert of Remove deprected functions from EncodedImageCallback and RtpRtcp (patchset #4 id:100001 of https://codereview.webrtc.org/2405173006/ )
>
> Reason for revert:
> This might be breaking projects downstream.
>
> Original issue's description:
> > Remove deprected functions from EncodedImageCallback and RtpRtcp
> >
> > Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> > These methods should no longer be used anywhere and it's safe to remove
> > them.
> >
> > BUG=chromium:621691
> >
> > Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> > Cr-Commit-Position: refs/heads/master@{#14902}
>
> TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:621691
>
> Committed: https://crrev.com/6c78307a21252c2dbd704f6d5e92a220fb722ed4
> Cr-Commit-Position: refs/heads/master@{#14914}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2467373003
Cr-Commit-Position: refs/heads/master@{#14915}
Reason for revert:
This might be breaking projects downstream.
Original issue's description:
> Remove deprected functions from EncodedImageCallback and RtpRtcp
>
> Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
> These methods should no longer be used anywhere and it's safe to remove
> them.
>
> BUG=chromium:621691
>
> Committed: https://crrev.com/fa565842718ad178a7562721b25d916fbabc2b92
> Cr-Commit-Position: refs/heads/master@{#14902}
TBR=mflodman@webrtc.org,stefan@webrtc.org,sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2474433008
Cr-Commit-Position: refs/heads/master@{#14914}
The H264SpsPpsTracker class:
- Keeps track of all received SPS/PPS.
- Decides whether a packet should be inserted into the PacketBuffer or not.
- Don't insert if this packet only contains SPS and/or PPS.
- Don't insert if this is the first packet of and IDR and we have not
received the required SPS/PPS.
- Insert start codes, and in the case of the first packet of an IDR prepend
the bitstream with the given SPS/PPS for this IDR.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2466993003
Cr-Commit-Position: refs/heads/master@{#14906}
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
Review-Url: https://codereview.webrtc.org/2468753002
Cr-Original-Commit-Position: refs/heads/master@{#14880}
Cr-Commit-Position: refs/heads/master@{#14903}
Removed EncodedImageCallback::Encoded() and RtpRtcp::SendOutgoingData().
These methods should no longer be used anywhere and it's safe to remove
them.
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2405173006
Cr-Commit-Position: refs/heads/master@{#14902}
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:
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}
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}
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}
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}
