Commit graph

148 commits

Author SHA1 Message Date
philipel
e6542f2112 Removed unused include from encoded_image.h.
Bug: webrtc:9378
Change-Id: Ie26ab4d30d62ec109a8be638661789399821c162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179525
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31758}
2020-07-17 14:14:03 +00:00
Erik Språng
969ccf0e12 Reland "VP9 decoder: Sets thread count based on resolution, reinit on change."
This is a reland of d592575698

Patchset 2 is a reland of
https://webrtc-review.googlesource.com/c/src/+/177012

Patchset 3 is a fix for a potential crash when InitDecode()is called from
VideoStreamDecoderImpl::GetDecoder(), where the decoder_settings
parameter is a but surprisingly set to nullptr.

Original change's description:
> VP9 decoder: Sets thread count based on resolution, reinit on change.
>
> Previously, number of decoder threads for VP9 were always set to 8 but
> with a cap at number of cores. This was done since we "can't know" the
> resolution that will be used.
>
> With this change, we now intialize the number of threads based on
> resolution given in InitDecode(). If a resolution change happens in
> flight, it requires a keyframe. We therefore parse the header from
> any key frame and if it has a new resolution, we re-initialize the
> decoder.
>
> The number of threads used is based on pixel count. We set one thread
> as target for 1280x720, and scale up lineraly from there. The 8-thread
> cap is gone, but still limit it core count.
>
> This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.
>
> Bug: webrtc:11551
> Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31507}

Bug: webrtc:11551
Change-Id: I2b4b146d0b8319f07ce1660202d6aa4b374eb015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177246
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31527}
2020-06-15 19:14:52 +00:00
Erik Språng
ef377ec6d5 Revert "VP9 decoder: Sets thread count based on resolution, reinit on change."
This reverts commit d592575698.

Reason for revert: May cause crashes.

Original change's description:
> VP9 decoder: Sets thread count based on resolution, reinit on change.
> 
> Previously, number of decoder threads for VP9 were always set to 8 but
> with a cap at number of cores. This was done since we "can't know" the
> resolution that will be used.
> 
> With this change, we now intialize the number of threads based on
> resolution given in InitDecode(). If a resolution change happens in
> flight, it requires a keyframe. We therefore parse the header from
> any key frame and if it has a new resolution, we re-initialize the
> decoder.
> 
> The number of threads used is based on pixel count. We set one thread
> as target for 1280x720, and scale up lineraly from there. The 8-thread
> cap is gone, but still limit it core count.
> 
> This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.
> 
> Bug: webrtc:11551
> Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31507}

TBR=ilnik@webrtc.org,sprang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11551
Change-Id: Id235c8ded83b3e1fc1d132c8f56c9f20001f6f22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177242
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31521}
2020-06-15 10:11:09 +00:00
Erik Språng
d592575698 VP9 decoder: Sets thread count based on resolution, reinit on change.
Previously, number of decoder threads for VP9 were always set to 8 but
with a cap at number of cores. This was done since we "can't know" the
resolution that will be used.

With this change, we now intialize the number of threads based on
resolution given in InitDecode(). If a resolution change happens in
flight, it requires a keyframe. We therefore parse the header from
any key frame and if it has a new resolution, we re-initialize the
decoder.

The number of threads used is based on pixel count. We set one thread
as target for 1280x720, and scale up lineraly from there. The 8-thread
cap is gone, but still limit it core count.

This means for instance: 1 <= 720p, 2 for 1080p, 4 for 1440p, 9 for 4K.

Bug: webrtc:11551
Change-Id: I14c169a6c651c50bd1b870c4b22bc4495c8448fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174460
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31507}
2020-06-11 17:43:27 +00:00
Danil Chapovalov
f2c0f15282 In media/ and modules/video_coding replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: I5c7f5dc99e62619403ed726c23201ab4fbd37cbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175647
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31340}
2020-05-25 08:46:30 +00:00
Henrik Boström
012aa375b1 Asynchronous QualityScaler: Callback-based CheckQpTask.
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing

Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
  void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
      rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
  QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
      bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).

QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.

The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.

Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-28 09:00:15 +00:00
Mirko Bonadei
6415dcad7a Remove WebRTC-ExperimentalScreenshareSettings.
This field trial is unused.

Bug: webrtc:11503
Change-Id: Id79b0dc64fed3559b9b63ebcf539e5536ddad589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173339
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31090}
2020-04-16 18:15:08 +00:00
Evan Shrubsole
ce0a11d5f9 Unify AdaptationReason and AdaptReason enums.
Moves the unified AdaptationReason to the api/ folder.

Bug: webrtc:11392
Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31084}
2020-04-16 13:33:49 +00:00
Ilya Nikolaevskiy
ae92244054 Clean VP8 header parser
The old implementation has undefined behavior in it (unaligned read of uint32_t)
Now it's closer to the reference implementation:
https://tools.ietf.org/html/rfc6386#section-20.2

Also, added some comments and named some variables to make it more clear, that the
parser actually does.

Bug: chromium:1057551
Change-Id: I84c1912867e2794502e8a63302c938a0cbab2c4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169545
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30661}
2020-03-02 15:05:48 +00:00
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18 16:09:50 +00:00
Mirta Dvornicic
6799d732d5 Delete DefaultVideoBitrateAllocator.
It was removed from tests in https://webrtc-review.googlesource.com/c/src/+/123540.

If simulcast is not used, SimulcastRateAllocator returns the
same allocation as DefaultVideoBitrateAllocator.

Bug: webrtc:10164
Change-Id: I3d3e1aefe2fcc2bf853cd63c75e008b86eff9241
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168496
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30509}
2020-02-12 21:29:09 +00:00
Danil Chapovalov
5528402ef8 Use newer version of TimeDelta and TimeStamp factories in modules/
This change generated with following commands:
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find modules -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I117d64a54950be040d996035c54bc0043310943a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168340
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30489}
2020-02-10 11:49:57 +00:00
Henrik Boström
48258acabf [Overuse] Implement Resource and ResourceUsageListener.
The Resource interface (previously a skeleton not used outside of
testing) is updated to inform listeners of changes to resource
usage. Debugging methods are removed (Name, UsageUnitsOfMeasurements,
CurrentUsage). The interface is implemented by
OveruseFrameDetectorResourceAdaptationModule's inner classes
EncodeUsageResource and QualityScalerResource.

The new ResourceUsageListener interface is implemented by
OveruseFrameDetectorResourceAdaptationModule. In order to avoid adding
AdaptationObserverInterface::AdaptReason to the ResourceUsageListener
interface, the module figures out if the reason is "kCpu" or "kQuality"
by looking which Resource object triggered
OnResourceUsageStateMeasured(). These resources no longer need an
explicit reference to OveruseFrameDetectorResourceAdaptationModule and
could potentially be used by a different module.

In this CL, AdaptationObserverInterface::AdaptDown()'s return value is
still needed by QualityScaler. This is mirrored in the return value of
ResourceUsageListener::OnResourceUsageStateMeasured(). A TODO is added
to remove it and a comment explains how the current implementation
seems to break the contract of the method (as was the case prior to
this CL).

Follow-up work include:
- Move EncodeUsageResource and QualityScalerResource to separate files.
- Make resources injectable, allowing fake resources in testing and
  removing OnResourceOveruseForTesting() methods.
  (Investigate adding the necessary input signals to the Resource
  interface or relevant sub-interfaces so that the module does not need
  to know which Resource implementation is used.)
- And more! See whiteboard :)

Bug: webrtc:11222
Change-Id: I0a46ace4a2e617874e3ee97e67e3a199fef420a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30469}
2020-02-06 12:45:14 +00:00
Evan Shrubsole
73a5e916a9 Remove task_queue dependency for QualityScaler
This allows for the possiblity to move the QualityScaler
out of the VideoStreamEncoder in the future.


Bug: webrtc:11222
Change-Id: I1d563cf08791e27ff5065ce90bcb150a7974d868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167534
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30406}
2020-01-29 12:14:10 +00:00
Florent Castelli
907dc806c7 Reland "Add support for RtpEncodingParameters::max_framerate"
Perf test failure was fixed separately.

TBR=steveanton@webrtc.org,sprang@webrtc.org,asapersson@webrtc.org

Original change's description:
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

Bug: webrtc:11117
Change-Id: I9c1daf7887c2024c6669dc79bff89d737417458c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161445
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30030}
2019-12-06 15:11:54 +00:00
Florent Castelli
5cef9c3581 Revert "Add support for RtpEncodingParameters::max_framerate"
This reverts commit 15be5282e9.

Reason for revert: crbug.com/1028937

Original change's description:
> Add support for RtpEncodingParameters::max_framerate
> 
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
> 
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
2019-11-27 14:01:53 +00:00
Artem Titov
a101a4f186 Reland "Add IvfVideoFrameGenerator"
This is a reland of 712a26f384

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

Bug: webrtc:10138
Change-Id: If522d079f0a1e30d6f2b330792aa1d1fc043b8b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160418
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29913}
2019-11-26 09:56:47 +00:00
Steve Anton
13ea34f305 Revert "Add IvfVideoFrameGenerator"
This reverts commit 712a26f384.

Reason for revert: consistently failing on iOS64 Debug: https://ci.chromium.org/p/webrtc-internal/builders/ci/iOS64%20Debug/20119

Original change's description:
> Add IvfVideoFrameGenerator
> 
> Bug: webrtc:10138
> Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29902}

TBR=ilnik@webrtc.org,titovartem@webrtc.org

Change-Id: Ie34e254a7a4ff5ff8fdab7c6b3212792b52b6f53
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160560
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29910}
2019-11-25 21:58:02 +00:00
Florent Castelli
15be5282e9 Add support for RtpEncodingParameters::max_framerate
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.

Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
2019-11-25 16:43:59 +00:00
Artem Titov
712a26f384 Add IvfVideoFrameGenerator
Bug: webrtc:10138
Change-Id: Iea590f334d22fb7d22077c9bdd3b5ba79691df2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160185
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29902}
2019-11-25 13:38:09 +00:00
Artem Titov
5831ddad65 Introduce IVF file reader
Bug: webrtc:10138
Change-Id: I97d332942f4e645527330159efefb1cb1d8034a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160008
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29844}
2019-11-20 13:20:56 +00:00
Åsa Persson
e644a03195 Add field trial for rampup in quality based on available bandwidth.
Bug: none
Change-Id: I32e1ea6fb2f2e20fc631e09b02c8f3a11b6c9fac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158888
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29751}
2019-11-11 10:13:28 +00:00
Rasmus Brandt
2b9317ad76 Stop checking VP8BaseHeavyTl3RateAllocation field trial on every frame.
- Centralize field trial string reading to RateControlSettings
- Cache RateControlSettings at all production code use sites

Bug: None
Change-Id: I0dbce9cc97fea0bc780982e7ef270b417a8c15bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158664
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29680}
2019-11-04 13:50:59 +00:00
Danil Chapovalov
eb90e6ffe3 Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.

Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
2019-10-15 09:17:36 +00:00
Ilya Nikolaevskiy
5963c7cf0a Count disabled due to low bw streams or layers as bw limited quality in GetStats
Bug: webrtc:11015
Change-Id: I65cd890706f765366d89ded8c21fa7507797fc23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155964
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29421}
2019-10-09 16:58:34 +00:00
Ilya Nikolaevskiy
e93b1fe8fd Improve bitstream dumping logic to handle multiple SLs correctly
Before this change all layers were glued together at the receive side
into a single IVF frame. This confuses most bitstream parsers.
Since this change all spatial layers would be written as separate frames
on the receive side also (on the send side it's already done that way).

Bug: none
Change-Id: I68543e4d4b336f87699ec3b4a113b8c93af0b7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156082
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29401}
2019-10-08 11:55:19 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Erik Språng
2b9dba3d9c Implement stable rate support in SimulcastRateAllocator
Bug: webrtc:10126
Change-Id: I2ea8d27b0bd6f7ffd1ebbba451bd1ce1f2eee3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151121
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29097}
2019-09-06 11:17:30 +00:00
Jonas Olsson
9fd2908f2e Remove unused framerate parameter from simulcast bitrate allocator.
It's not removed from VideoBitrateAllocationParameters as that struct
is part of the API.

Bug: webrtc:9883
Change-Id: I69f683e3c1dc3a0edc1711f6289514b86b05ad77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149815
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28935}
2019-08-22 09:01:54 +00:00
Åsa Persson
f5e5d250bc BalancedDegradationSettings: add option to configure a min framerate diff.
If a framerate reduction (input fps - restricted fps) is less than the
configured diff, shorten interval to next qp check.

Bug: none
Change-Id: Ia0b9e0638e5ba75cdc20a1bb45bfcb7d858c5f89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149040
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28880}
2019-08-16 16:13:46 +00:00
Erik Språng
d7ee76cadd Wire up field trials for some experimental screenshare settings
Bug: b/132074409
Change-Id: I83d5334255bad4fcf585b9850506bbfe1914ba57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147868
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28758}
2019-08-05 13:47:01 +00:00
Florent Castelli
8bbdb5b9bd Update VideoBitrateAllocator allocate to take a struct with more fields
We want to evaluate more data in order to make better choices in the
bitrate allocators.
In order to freely update the parameter list without
breaking the API many times for projects customizing them, we'll use a
struct instead.

Bug: webrtc:10126
Change-Id: I443f86781c5134950294cdd1e3197a47447cf973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141418
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28748}
2019-08-02 13:52:54 +00:00
Florent Castelli
596ed251e1 Don't assume all simulcast screenshare have 2 temporal layers
The simulcast allocator would only set bitrates for the first 2 layers
in conference_screenshare_mode.
That would trigger an issue in the VP8 encoder initialization that expects
to have growing bitrates for the layers (3rd layer would have the same
bitrate as the 2nd one).

Bug: webrtc:8785
Change-Id: Ic6c940b78022387841b28074b373be6b2f45cb15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145922
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28598}
2019-07-18 11:51:03 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Niels Möller
825aad13dc Delete almost all includes of platform_file.h
Only remaining user is WavReader. Demote its constructor
accepting a PlatformFile to private, to refactor implementation
in a later cl.

Bug: webrtc:6463
Change-Id: I7b950be6f02073cb135dd0fab1190b9dc0de1fba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144025
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28410}
2019-06-28 07:30:15 +00:00
Johnny Lee
c53850675b Enable H.264 temporal scalability in simulcast.
Bug: webrtc:10651
Change-Id: I58372186930ce33e925f85edb0f308657dbfe273
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142840
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28381}
2019-06-26 09:21:32 +00:00
Åsa Persson
1231419785 BalancedDegradationSettings: Add option to configure QP thresholds.
Add possibility to configure low/high QP thresholds based on resolution.

Bug: none
Change-Id: Iaa3168b77678bd74feb67295d7658c0140721231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141867
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28348}
2019-06-24 09:32:51 +00:00
Niels Möller
4d504c76cb New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer
Bug: webrtc:9378
Change-Id: I62b7adbd9dd539c545b5b1b1520721482a4623c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138820
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28317}
2019-06-19 07:02:34 +00:00
Elad Alon
370f93a34a Reland "Inform VideoEncoder of negotiated capabilities"
This is a reland of 11dfff0878

Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org

Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
2019-06-11 14:49:37 +00:00
Philip Eliasson
49d661a7d3 Revert "Inform VideoEncoder of negotiated capabilities"
This reverts commit 11dfff0878.

Reason for revert: Downstream import failure.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
> 
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
> 
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
2019-06-11 11:56:04 +00:00
Elad Alon
11dfff0878 Inform VideoEncoder of negotiated capabilities
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().

Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
2019-06-11 11:32:13 +00:00
Elad Alon
fb0878197e Reland "Refactor handling of configuration overrides from Vp8FrameBufferController"
This is a reland of 4d6795f828

Original change's description:
> Refactor handling of configuration overrides from Vp8FrameBufferController
>
> Make Vp8FrameBufferController::UpdateConfiguration return a set
> of desired overrides. These overrides are cumulative with
> previously returned override sets.
>
> Bug: webrtc:10382
> Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27835}

TBR=brandtr@webrtc.org,sprang@webrtc.org

Bug: webrtc:10382
Change-Id: I0b313d19843edf429aeeaa4deafc5426c434a0fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135942
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27904}
2019-05-10 08:44:02 +00:00
Elad Alon
3d622d6e5c Revert "Refactor handling of configuration overrides from Vp8FrameBufferController"
This reverts commit 4d6795f828.

Reason for revert: chromium:961253

Original change's description:
> Refactor handling of configuration overrides from Vp8FrameBufferController
>
> Make Vp8FrameBufferController::UpdateConfiguration return a set
> of desired overrides. These overrides are cumulative with
> previously returned override sets.
>
> Bug: webrtc:10382
> Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27835}

TBR=brandtr@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org


Bug: chromium:961253
Change-Id: I06f0eafd4f38c441ddbdfeebae8055b02465eb9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135940
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27900}
2019-05-09 21:09:23 +00:00
Åsa Persson
517678cc49 Add ability to configure quality scaler settings through field trial.
optional<int> min_frames: The minimum number frames to observe to make a
                          scaling decision.
Default: kMinFramesNeededToScale in quality_scaler.cc

optional<double> initial_scale_factor: The sample period scale factor.
Default: kSamplePeriodScaleFactor in quality_scaler.cc

optional<double> scale_factor: Option to use a reduced sampling interval when
                               last check did not result in an adaptation (if
                               unset the initial_scale_factor is used).

Bug: none
Change-Id: I3bb955d1f8d7d7d49bc118361614b5aa59605231
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135125
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27860}
2019-05-06 13:20:27 +00:00
Elad Alon
4d6795f828 Refactor handling of configuration overrides from Vp8FrameBufferController
Make Vp8FrameBufferController::UpdateConfiguration return a set
of desired overrides. These overrides are cumulative with
previously returned override sets.

Bug: webrtc:10382
Change-Id: I1aa9544ae0cf6c57115e80963b3bbcdc3101db5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134649
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27835}
2019-05-03 09:12:56 +00:00
Elad Alon
979c4426a4 Rename "UpdateLayerConfig" to "NextFrameConfig"
Rename "UpdateLayerConfig" to the more appropriate "NextFrameConfig".
Also update some comments in vp8_frame_buffer_controller.h.

Bug: None
Change-Id: Iba8227f84e33e5ebd28d2eeb10fe03e776036603
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133202
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27660}
2019-04-17 11:59:07 +00:00
Erik Språng
16cb8f5d74 Reland "Replace usage of old SetRates/SetRateAllocation methods"
This is a reland of 7ac0d5f348

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org

Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
2019-04-12 13:37:32 +00:00
Niels Möller
7aacdd9515 Reland "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
This is a reland of 39d3a7de02

Original change's description:
> Delete CodecSpecificInfo argument from VideoDecoder::Decode
>
> Bug: webrtc:10379
> Change-Id: I079b419604bf4e9c1994fe203d7db131a0ccddb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125920
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27022}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10379
Change-Id: I8197bebd2ae7dc460644a98795b8257b033c27c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126480
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27565}
2019-04-11 13:03:52 +00:00
Minyue Li
7ddef1af88 Revert "Replace usage of old SetRates/SetRateAllocation methods"
This reverts commit 7ac0d5f348.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
> 
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
> 
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}

TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
2019-04-11 10:50:29 +00:00
Erik Språng
7ac0d5f348 Replace usage of old SetRates/SetRateAllocation methods
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.

Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
2019-04-11 07:46:09 +00:00