One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
The delay_ms stat in AudioprocessStats should be an Optional, because its value is not always computed. This CL changes it to an optional.
Bug: webrtc:8569
Change-Id: I42fd7a86b975c766b685444bf1829511f790da2a
Reviewed-on: https://webrtc-review.googlesource.com/33320
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21293}
The title of Dock window on OSX10.12 is still 'Dock' with layer number 20.
So the removed codes here is reduntant.
Also fix a wrong comment.
Bug: webrtc:8460
Change-Id: I72d4c8f5741a1ccb00aa45897f11e85af8d24e05
Reviewed-on: https://webrtc-review.googlesource.com/33123
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21287}
This reverts commit 59283e4c66d038a00923736685457f4b53f922fe.
Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.
Sample error log:
[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
>
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
>
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
>
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
Also cleaned up a bit in RtpFrameReferenceFinder.
Bug: chromium:762556
Change-Id: Ib08d2e7ce4b146b359ce9ba823f3aa15776c71bc
Reviewed-on: https://webrtc-review.googlesource.com/32301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21282}
This CL removes the following GN variables: rtc_build_libyuv,
rtc_libyuv_dir (as requested in webrtc:7906).
It also removes some unneeded dependencies on //third_party/libyuv.
WebRTC targets were using public_deps to depend on //third_party/libyuv
and this created a build graph where targets that were depending on
//third_party/libyuv were not declaring the dependency to GN because
they were somehow getting it from another target that was exposing
//third_party/libyuv header files even if it wasn't directly depending
on it.
Bug: webrtc:8605, webrtc:7906
Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb
Reviewed-on: https://webrtc-review.googlesource.com/32201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21275}
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Though hacky, this is very similar to currently used implementation
in the RTCPSender::SendFeedbackPacket
Bug: webrtc:8239
Change-Id: I237b422ae1594dede78cb63daa4aa42b6774d6fe
Reviewed-on: https://webrtc-review.googlesource.com/32680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21274}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
in particular change bitrate type to int64_t to follow style guide.
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Bug: webrtc:8239
Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba
Reviewed-on: https://webrtc-review.googlesource.com/32302
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21250}
Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.
Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.
Bug: webrtc:8643
Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21249}
FFMpeg H264 decoder uses YUVJ420 when video_full_range_flag=1 in
bitstream.
Information about color range might be useful for color converter
and renderer. But currently there is no way to extract it from
the wrapper.
Bug: webrtc:8185
Change-Id: Ifd1113f0eee3d7b5906d0cefbc29b4a1061262f6
Reviewed-on: https://webrtc-review.googlesource.com/32000
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21246}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
This tool is used downstream, so we want to christen rtc_tools as
a kind of api dir for tools. Tools in other locations should be
considered off limits.
I chose rtc_tools because video_quality_toolchain is already there,
which is also used downstream.
Bug: None
Change-Id: I234d874c8a590ca7413357ecda26b16d9b399836
Reviewed-on: https://webrtc-review.googlesource.com/32340
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21236}
The problem was that the encoder was feeded with frames that had 0 as
a timestamp. This confused the encoder. H264 high profile support
clause was also wrong and is corrected.
Bug: webrtc:8601
Change-Id: Ic5a893b4b7573e694f865b63620843b2c9aa489f
Reviewed-on: https://webrtc-review.googlesource.com/32300
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21234}
1. Added WEBRTC_FUCHSIA define.
2. Added PlatformThreadId typedef for Fuchsia.
3. Updated ifdefs for _strnicmp()/strncasecmd(), so _strnicmp()
is used on all platforms
3. Updated ifdefs in clock.cc to avoid invalid assumption that
POSIX = LINUX || MAC .
Bug: chromium:750940
Change-Id: Id7aa98e017f467bcebb78a0b298ba91655502072
Reviewed-on: https://webrtc-review.googlesource.com/31641
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21233}
When calling CGDisplayStreamCreate(properties = nullptr) this
causes kCGDisplayStreamShowCursor to default to kCFBooleanTrue.
This CL set it to false always as it was assumed. Also if true
this causes some lags when moving the mouse pointer on the capture
side and in any case webrtc::MouseCursorMonitorMac already implements
a custom way to capture the mouse. Which appears to be more efficient
in this usecase.
Bug: webrtc:8625
Change-Id: Id0fae38fa47503d87d1890213706149762fa67fb
Reviewed-on: https://webrtc-review.googlesource.com/30902
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21231}
Concealment is never used in WebRTC since we never feed decoders with
broken bitstream. If so, there is no need to evaluate concealment
quality.
But if we still want to evaluate it then the tests should be
redesigned: recovery frames should be generated with reasonable
interval and quality thresholds should be set to acceptable level.
Bug: webrtc:8524
Change-Id: Ie7197e0a5a88aafcb3b2698185edcb43b71fae3b
Reviewed-on: https://webrtc-review.googlesource.com/32303
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21230}
In order to support WinUWP platform, all main(..) routines must be normalized to the formal int main(int argc, char* argv[]) form. A platform wrapper main is auto-created linking against the default main(...). This can only work if the linkage is exactly matching the proper formal definition and not a loosely defined main(...) alternative.
Bug: webrtc:8608
Change-Id: I606663aaea7df1792c7c5636279617b8926fa5cc
Reviewed-on: https://webrtc-review.googlesource.com/28721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21229}
Avoids hitting a DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
when first init attempt has failed and we try again.
Bug: b/69434512
Change-Id: I4396ba22981d9258d6d72188bad66104255f19cf
Reviewed-on: https://webrtc-review.googlesource.com/31842
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21218}
This CL modifies the refactored render buffering scheme in AEC3
so that:
-A non-causal state can never occur which means that situations with
nonrecoverable echo should not occur.
-For a stable audio pipeline with a predefined API call jitter,
render overruns and underruns can never occur.
Bug: webrtc:8629,chromium:793305
Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145
Reviewed-on: https://webrtc-review.googlesource.com/29861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21215}
The experimental function that scales the histogram of inter-arrival times in NetEq suffered from an overflow bug. This caused unexpected increases in the calculated target level.
Bug: webrtc:8381
Change-Id: I2af4d22119fdc684b3cac838c9b317959af17a1f
Reviewed-on: https://webrtc-review.googlesource.com/30261
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21213}
- Defines stereo codec case, similar to RTX, that adds stereo codec to the SDP
negotiation. The underlying codec's payload type is similarly defined by "apt".
- If this negotiation is successful, codec name is included in sdp line via
"acn".
- Adds codec setting initializers for these specific stereo cases.
- Introduces new Stereo*Factory classes as optional convenience wrappers that
inserts stereo codec to the existing set of supported codecs on demand.
This CL is the step 5 for adding alpha channel support over the wire in webrtc.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ie12c56c8fcf7934e216135d73af33adec5248f76
Reviewed-on: https://webrtc-review.googlesource.com/22901
Commit-Queue: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21210}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
The computation (x-127) << 8 is undefined for x < 127.
This CL replaces the shift with a multiplication: (x-127) * (1 << 8)
Bug: chromium:793201
Change-Id: I38b40bd88300208a0bfbbd8fe144b0a5b51a48ed
Reviewed-on: https://webrtc-review.googlesource.com/31800
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21205}
Arguably we should add a few more targets, for instance a utility
target, but I tried to create as few targets as possible here based on
the current usage.
Bug: webrtc:6828
Change-Id: Icc9400d76f4016c8b0943aa734430955208a14f8
Reviewed-on: https://webrtc-review.googlesource.com/28301
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21199}
The definition of this field in RFC 3550 says that under certain
conditions it may have a negative value. This change exposes that
property in the WebRTC API.
Bug: webrtc:8626
Change-Id: I4ee249da045dcee940db66ebd915268a97fc13db
Reviewed-on: https://webrtc-review.googlesource.com/31260
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21159}
Instead of modifying the API, we'll add a new function to return
the true value, and have a shim that returns what other code expects.
> This reverts commit 4c34f435db2b921b82b8be19ee5c1746f46cb188.
>
> Reason for revert: Broke internal projects. Type mismatch.
>
> Original change's description:
> > Make RTCP cumulative_lost be a signed value
> >
> > This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> > See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> >
> > Noticed on discuss-webrtc mailing list.
> >
> > BUG=webrtc:8626
> >
> > Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> > Reviewed-on: https://webrtc-review.googlesource.com/30901
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21142}
>
> TBR=stefan@webrtc.org,hta@webrtc.org
>
> Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8626
> Reviewed-on: https://webrtc-review.googlesource.com/31040
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21144}
Change-Id: I95c8c248f4f85c4d1aa2a47424d8c4d954d4ae7a
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21154}
This reverts commit 4c34f435db2b921b82b8be19ee5c1746f46cb188.
Reason for revert: Broke internal projects. Type mismatch.
Original change's description:
> Make RTCP cumulative_lost be a signed value
>
> This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> See RFC 3550 Appendix A.3 for the reason why it may turn negative.
>
> Noticed on discuss-webrtc mailing list.
>
> BUG=webrtc:8626
>
> Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> Reviewed-on: https://webrtc-review.googlesource.com/30901
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21142}
TBR=stefan@webrtc.org,hta@webrtc.org
Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21144}
This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
See RFC 3550 Appendix A.3 for the reason why it may turn negative.
Noticed on discuss-webrtc mailing list.
BUG=webrtc:8626
Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
Reviewed-on: https://webrtc-review.googlesource.com/30901
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21142}
