When Chromium hooks up with the stereo codec, then it has difficulty
communicating with a google chrome without stereo codec. By design, we
do allow codec choice for the standalone codecs, but the problem is
that we do not handle the payload correctly, and thus the existence
of stereo codec will remove the payload registry of the standalone
version of its associated codec. (For example, stereo codec on top of
VP9 will remove the payload registry of standalone VP9 codec.)
This CL fixes the issue. When generating payload data, we should use
"stereo" as payload name, instead of its associated codecs.
Bug: webrtc:8657
Change-Id: I9e0b54de6bd41d370b9353f9553c998e4049789f
Reviewed-on: https://webrtc-review.googlesource.com/33122
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21523}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
This reverts commit 727b7d0470c0515397d21698ee089197c31cb5ff.
Reason for revert: Breaks build
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I8a0621eb91f9ce4835f012e74b6a1da9bf740963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/36940
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21465}
This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26
Original change's description:
> Reland "Put internal video codec factories into separate target"
>
> This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> Original change's description:
> > Put internal video codec factories into separate target
> >
> > The purpose is to start splitting out the dependencies to the built-in
> > SW video codecs, so that clients can decide to not depend on them and
> > get a reduction in binary size.
> >
> > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> >
> > Bug: webrtc:7925
> > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21381}
>
> Bug: webrtc:7925
> Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> Reviewed-on: https://webrtc-review.googlesource.com/35261
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21389}
Bug: webrtc:7925
Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
Reviewed-on: https://webrtc-review.googlesource.com/35501
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21464}
This reverts commit 718d8631b0294a8bdc56366b68c51e2f04cd0c9e.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback""
>
> This reverts commit 53d901332c2eb43cad0da5768c6f7a8c4aeb9590.
>
> Reason for revert: root cause has been found and will be addressed in the patch.The root cause was protection_bitrate_calculator_ is now destructed before worker_queue_, and worker_queue_ may contain tasks which involves protection_bitrate_calculator_, so they need to be destructed in the opposite order.
> That was not an issue since before this cl we didn't allocate protection_bitrate_calculator_ on the heap.
>
> Original change's description:
> > Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"
> >
> > This reverts commit e58e91b6d143ef847f8df24b19de4ba98cdb6f72.
> >
> > Reason for revert: Breaks downstream project b/70848177
> >
> > Original change's description:
> > > Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback
> > >
> > > Bug: webrtc:8656
> > > Change-Id: Iab4f6ab8997cb082762218afc8580e9985ac2522
> > > Reviewed-on: https://webrtc-review.googlesource.com/33010
> > > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21348}
> >
> > TBR=stefan@webrtc.org,philipel@webrtc.org,yinwa@webrtc.org
> >
> > Change-Id: Ic186ba78be429bd1046ceac15051a3382b6ffc4f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8656
> > Reviewed-on: https://webrtc-review.googlesource.com/35080
> > Commit-Queue: Lu Liu <lliuu@webrtc.org>
> > Reviewed-by: Lu Liu <lliuu@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21374}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
>
> Change-Id: Ie2b5a2a2ead0f20ac67c1ea9b8d192af66bddf8d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8656
> Reviewed-on: https://webrtc-review.googlesource.com/35320
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ying Wang <yinwa@webrtc.org>
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21409}
TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
Change-Id: I9773aaa942054dcfbab6002a5d713ab3526b0534
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8656
Reviewed-on: https://webrtc-review.googlesource.com/35700
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21410}
This reverts commit 53d901332c2eb43cad0da5768c6f7a8c4aeb9590.
Reason for revert: root cause has been found and will be addressed in the patch.The root cause was protection_bitrate_calculator_ is now destructed before worker_queue_, and worker_queue_ may contain tasks which involves protection_bitrate_calculator_, so they need to be destructed in the opposite order.
That was not an issue since before this cl we didn't allocate protection_bitrate_calculator_ on the heap.
Original change's description:
> Revert "Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback"
>
> This reverts commit e58e91b6d143ef847f8df24b19de4ba98cdb6f72.
>
> Reason for revert: Breaks downstream project b/70848177
>
> Original change's description:
> > Add ProtectionBitrateCalculator as an abstract class. ProtectionBitrateCalculatorDefault implements ProtectionBitrateCalculator. Register VideoSendStream to packet feedback
> >
> > Bug: webrtc:8656
> > Change-Id: Iab4f6ab8997cb082762218afc8580e9985ac2522
> > Reviewed-on: https://webrtc-review.googlesource.com/33010
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21348}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org,yinwa@webrtc.org
>
> Change-Id: Ic186ba78be429bd1046ceac15051a3382b6ffc4f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8656
> Reviewed-on: https://webrtc-review.googlesource.com/35080
> Commit-Queue: Lu Liu <lliuu@webrtc.org>
> Reviewed-by: Lu Liu <lliuu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21374}
TBR=stefan@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org,yujo@chromium.org,yinwa@webrtc.org
Change-Id: Ie2b5a2a2ead0f20ac67c1ea9b8d192af66bddf8d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8656
Reviewed-on: https://webrtc-review.googlesource.com/35320
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21409}
This reverts commit 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26.
Reason for revert: Breaking internal builds
Original change's description:
> Reland "Put internal video codec factories into separate target"
>
> This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> Original change's description:
> > Put internal video codec factories into separate target
> >
> > The purpose is to start splitting out the dependencies to the built-in
> > SW video codecs, so that clients can decide to not depend on them and
> > get a reduction in binary size.
> >
> > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> >
> > Bug: webrtc:7925
> > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21381}
>
> Bug: webrtc:7925
> Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> Reviewed-on: https://webrtc-review.googlesource.com/35261
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21389}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: I8d3b788cc9e43261b3ed6d3d52427b5e26bc827e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/35187
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21393}
This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
Original change's description:
> Put internal video codec factories into separate target
>
> The purpose is to start splitting out the dependencies to the built-in
> SW video codecs, so that clients can decide to not depend on them and
> get a reduction in binary size.
>
> Replaces https://webrtc-review.googlesource.com/c/src/+/29101
>
> Bug: webrtc:7925
> Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> Reviewed-on: https://webrtc-review.googlesource.com/33420
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21381}
Bug: webrtc:7925
Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
Reviewed-on: https://webrtc-review.googlesource.com/35261
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21389}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
This reverts commit 51698aefd4925f2dfa0310a321f836d433fa9258.
Reason for revert: Breaks builds because badly formatted deps
Original change's description:
> Put internal video codec factories into separate target
>
> The purpose is to start splitting out the dependencies to the built-in
> SW video codecs, so that clients can decide to not depend on them and
> get a reduction in binary size.
>
> Replaces https://webrtc-review.googlesource.com/c/src/+/29101
>
> Bug: webrtc:7925
> Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> Reviewed-on: https://webrtc-review.googlesource.com/33420
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21381}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: Ib85f77fea756f4beb6a95b45cb132cbdc424ef00
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/35260
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21383}
The purpose is to start splitting out the dependencies to the built-in
SW video codecs, so that clients can decide to not depend on them and
get a reduction in binary size.
Replaces https://webrtc-review.googlesource.com/c/src/+/29101
Bug: webrtc:7925
Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
Reviewed-on: https://webrtc-review.googlesource.com/33420
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21381}
This reverts commit 28a06b16cc4daa9f380ad45af8acfd11b6057283.
Reason for revert: Causes some unexpected perf changes.
Original change's description:
> Smoother frame dropping when screenshare_layers limits fps
>
> Currently, when input fps is higher than the configured target fps in
> screenshare_layers, we drop frames with the help of a rate tracker using
> a one second sliding window. This is not optimal.
> For instance, given a 5fps limit and a 30fps capturer, the window will
> not be saturated until we have added the first 5 frames. Then we will
> drop all frames until the oldest one drops out, at which point we can
> immediately fill it's place. This results in quick 5 frame bursts every
> second.
>
> In addition to rate tracker, also set a limit on minimum interval
> required between input frames, based on target frame rate.
>
> Bug: webrtc:4172
> Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
> Reviewed-on: https://webrtc-review.googlesource.com/34380
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21325}
TBR=ilnik@webrtc.org,sprang@webrtc.org
Change-Id: I7ca5b0c847310dbb11dce28773082eac946c0ba4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4172
Reviewed-on: https://webrtc-review.googlesource.com/34780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21354}
Currently, when input fps is higher than the configured target fps in
screenshare_layers, we drop frames with the help of a rate tracker using
a one second sliding window. This is not optimal.
For instance, given a 5fps limit and a 30fps capturer, the window will
not be saturated until we have added the first 5 frames. Then we will
drop all frames until the oldest one drops out, at which point we can
immediately fill it's place. This results in quick 5 frame bursts every
second.
In addition to rate tracker, also set a limit on minimum interval
required between input frames, based on target frame rate.
Bug: webrtc:4172
Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
Reviewed-on: https://webrtc-review.googlesource.com/34380
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21325}
TBR=brandtr@webrtc.org,stefan@webrtc.org
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: I553cea30dcda34b753b5224f15094a1b7b70a750
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-Original-Commit-Position: refs/heads/master@{#21249}
Reviewed-on: https://webrtc-review.googlesource.com/33360
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21319}
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}
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}
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}
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}
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}
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}
- 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}
In the ClearTo function we risked removing one packet too many from the set of
|missing_packets_|, and in the case of H264 this would cause us to create incomplete
delta frames. This is turn disrupt the stream and a keyframe request has to be made.
Bug: webrtc:8536
Change-Id: Ie7ccd5f1631a4cf3bd463878d5b0fe746744ec23
Reviewed-on: https://webrtc-review.googlesource.com/30141
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21119}
WebRTC standalone tests show 24% speed up on foreman_cif, 16% speed up
on a 240p clip and 11% speed up on Bridge_r180.
Bug: None
Change-Id: I433b7a8841bd9df2402575f72dd1984cc5e011a9
Reviewed-on: https://webrtc-review.googlesource.com/29260
Commit-Queue: Jerome Jiang <jianj@google.com>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21096}
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.
Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
This will also cause us to use the new Android HardwareVideoEncoder,
instead of the deprecated MediaCodecVideoEncoderFactory. Unfortunately,
the new HW encoder does not seem to work as good as the old (or the new
encoder is more strict with return values or something). I don't think
it adds much value to continue testing the deprecated encoder, so I
filed a bug for fixing the new encoder, and in this CL I disabled the
tests on Android. I want to remove as many places as possible where we
use the old WebRtcVideoEncoderFactory interface, because it makes it
more difficult to migrate to the new interface.
Bug: webrtc:7925
Change-Id: If8e34752148a5e5139944d2dfbe7e231fe58aeb9
Reviewed-on: https://webrtc-review.googlesource.com/27540
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21037}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=sprang@webrtc.org
Bug: None
Change-Id: Ic429f28a8610ca798e29c45ec1f64604d6f9687f
Reviewed-on: https://webrtc-review.googlesource.com/23603
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20955}
This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1.
Reason for revert: Breaks downstream project.
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org
Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.
This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT
Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Iedebf4dc56a973306e7d7e7649525879808dc72b
Reviewed-on: https://webrtc-review.googlesource.com/23578
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20878}
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.
On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.
Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
Because some hardware encoders may not preserve capture timestamp, use
rtp timestamp as a key to capture time storage and restore corrupted
capture timestamps whenever possible.
Bug: webrtc:8497
Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4
Reviewed-on: https://webrtc-review.googlesource.com/23620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20732}
