Commit Graph

1474 Commits

Author SHA1 Message Date
8c1bf9595a Reland "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad.

Reason for revert: downstream project fixed.

Original change's description:
> Revert "Add initial support for RtpEncodingParameters max_framerate."
>
> This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Add initial support for RtpEncodingParameters max_framerate.
> >
> > Add support to set the framerate to the maximum of |max_framerate|.
> > Different framerates are currently not supported per stream for video.
> >
> > Bug: webrtc:9597
> > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> > Reviewed-on: https://webrtc-review.googlesource.com/92392
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24270}
>
> TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9597
> Reviewed-on: https://webrtc-review.googlesource.com/94060
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24277}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Bug: webrtc:9597
Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e
Reviewed-on: https://webrtc-review.googlesource.com/100080
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:06:33 +00:00
941a07cca3 Remove all remaining non-test uses of std::stringstream.
Bug: webrtc:8982
Change-Id: I635a8545c46dc8c89663d64af351e22e65cbcb33
Reviewed-on: https://webrtc-review.googlesource.com/98880
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24715}
2018-09-13 08:52:05 +00:00
7d687b13ed SimulcastEncoderAdapter, don't start streams without enough bitrate
Currently a bug in InitEncode() sets all stream initially to active.
This CL actually bases the active-flag on available start bitrate.

Bug: webrtc:9747
Change-Id: If197b0c69376d96c717f2a391fba8108895018f3
Reviewed-on: https://webrtc-review.googlesource.com/99960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24711}
2018-09-12 16:34:45 +00:00
72bc8d6df6 Make the rtp timestamp member of EncodedImage private
A followup to https://webrtc-review.googlesource.com/c/src/+/82160,
which added accessor methods.

Bug: webrtc:9378
Change-Id: Id3cff46cde3a5a3fb6d6edd4e8dac26193e6481c
Reviewed-on: https://webrtc-review.googlesource.com/95103
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24705}
2018-09-12 13:44:36 +00:00
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
bfd412ef71 Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
This change passes a pointer (non-owning) down to the MediaChannel when set
in the RtpSender / RtpReceiver. This currently is not used to encrypt frames.

Bug: webrtc:9681
Change-Id: I385fa8b948427803cd3f9cef918c31d7754d1b4f
Reviewed-on: https://webrtc-review.googlesource.com/97000
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24694}
2018-09-11 20:10:44 +00:00
cb7e1d2edb Use SdpVideoFormat in VideoReceiveStream::Decoder
Replaces payload_name and codec_params.

Tbr: srte@webrtc.org
Bug: webrtc:9106
Change-Id: Ib45c501c6eb41e92fbb24ab00ada18bf10be42ed
Reviewed-on: https://webrtc-review.googlesource.com/98161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24691}
2018-09-11 15:03:04 +00:00
7cd0e15faf Add nisse@, ilnik@ and sprang@ as OWNERS to media/.
Bug: None
Change-Id: Id717b808749f44cfe4579faafcaf52d12ae6e8eb
Reviewed-on: https://webrtc-review.googlesource.com/99560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24684}
2018-09-11 12:21:48 +00:00
661a0c6b02 Updated min bitrate for high-quality screenshare simulcast stream
The min bitrate is too low, and burstiness may cause overuse when first
enabling the stream, if the total available bitrate is low.

Bug: webrtc:9734
Change-Id: I399e0e809648f064feb87c73ece0c23a569b2750
Reviewed-on: https://webrtc-review.googlesource.com/99506
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24678}
2018-09-11 11:27:08 +00:00
59ab3536a4 Add receive stream id argument to CreateDecoder() method
This is necessary to migrate some clients so that we can move forward
with removal of cricket::WebRtcVideoDecoderFactory.

TBR=stefan@webrtc.org

Bug: webrtc:7925
Change-Id: Icc2949e3f7f3137d1b68eb30874f14a33168e41f
Reviewed-on: https://webrtc-review.googlesource.com/97500
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24671}
2018-09-11 08:47:04 +00:00
04255172b6 Remove double declaration of 2 conversion functions.
The declaration in common_types.h is probably a left-over from a
previous cleanup.

Bug: None
Change-Id: I3ee1bad2494ede0022c6aa8fdd106035471d50e2
Reviewed-on: https://webrtc-review.googlesource.com/99220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24666}
2018-09-11 06:00:05 +00:00
7461eff1bd For simulcast screenshare, make 2 tl default for high stream.
Bug: webrtc:9734
Change-Id: I00400782686296b191f0f7a10a65f99253bea929
Reviewed-on: https://webrtc-review.googlesource.com/99101
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24642}
2018-09-10 09:07:59 +00:00
2e4419e05b Add option to only request a frame interval change via OnOutputFormatRequest.
OnOutputFormatRequest(const absl::optional<VideoFormat>& format)
changed to
OnOutputFormatRequest(
      const absl::optional<std::pair<int, int>>& target_aspect_ratio,
      const absl::optional<int>& max_pixel_count,
      const absl::optional<int>& max_fps)

Decouples:
- Resolution and fps requests.
- Resolution requests from aspect ratio requests.

Bug: webrtc:9597
Change-Id: I6f44c91283cf5474c6531e55773d2257e2341063
Reviewed-on: https://webrtc-review.googlesource.com/95423
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24623}
2018-09-07 10:36:27 +00:00
ed425d12c4 Multiplex codec cleanups
This CL performs some cleanups on multiplex files:
- Adds more comments to factory about usage.
- Moves image packer outside /include as it doesn't need to be public.
- Other small lint issues.

Bug: webrtc:9632
Change-Id: I2e2e6929ea13645aee5483a3697199d1e6184b32
Reviewed-on: https://webrtc-review.googlesource.com/98700
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24615}
2018-09-06 23:29:15 +00:00
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
1beef1a97a Delete VideoSendStream::EnableEncodedFrameRecording.
Use in VideoQualityTest replaced by creating a wrapper for the encoder.

Bug: None
Change-Id: I5c5519e147ca7ddb97696b0d6958a8a1f5cc6e83
Reviewed-on: https://webrtc-review.googlesource.com/94152
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24533}
2018-09-03 13:06:32 +00:00
d3b8c63b58 Reland "Add spatial index to EncodedImage."
This is a reland of da0898dfae3b0a013ca8ad3828e9adfdc749748d

Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}

Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
2018-08-31 07:35:52 +00:00
cc22f51988 Removing the intelligibility enhancer.
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).

Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
2018-08-30 21:29:57 +00:00
5a998d7246 Revert "Add spatial index to EncodedImage."
This reverts commit da0898dfae3b0a013ca8ad3828e9adfdc749748d.

Reason for revert: Broke downstream tests.

Original change's description:
> Add spatial index to EncodedImage.
> 
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
> 
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}

TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,srte@webrtc.org

Change-Id: Idb4fb9d72e5574d7353c631cb404a1311f3fd148
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9378
Reviewed-on: https://webrtc-review.googlesource.com/96664
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24486}
2018-08-29 14:36:05 +00:00
da0898dfae Add spatial index to EncodedImage.
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.

Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
2018-08-29 13:50:17 +00:00
c30a13147c Add experiment for boosted qp at lowest stream for screenshare
Bug: webrtc:9659
Change-Id: I2320afc393d6a78ae03a4f447f0e3333dd5748c4
Reviewed-on: https://webrtc-review.googlesource.com/95943
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24441}
2018-08-27 10:04:50 +00:00
1946a3f0fe Add frame rate parameter to SpatialLayer struct.
This will allow us to configure VP9 encoder to produce spatial layers
with different frame rates.

Bug: webrtc:9650
Change-Id: I3a9c58072003b8a8da681d5291d8f7ede7f52fa4
Reviewed-on: https://webrtc-review.googlesource.com/95427
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24435}
2018-08-26 19:19:36 +00:00
3288168e4f Enable video adaptation for all screenshare content
Since screenshare uses "Maintain resolution" degradation preference,
adapting should be enabled to reduce framerate if encoder can't keep up.

Bug: chromium:690537
Change-Id: I1f4418b7b7b4faa13f34d5353e3c625a59975c05
Reviewed-on: https://webrtc-review.googlesource.com/95460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24426}
2018-08-24 12:02:03 +00:00
3df1d5d2fb Revert removal of simulcast screenshare experimental code (killswitch checks)
This reverts commit a3df0f2d05c7b0973c31fe171507e97e588671a5.

Reason for revert: We decided to keep a killswitch in M70 just in case.

Original reviewed at: https://webrtc-review.googlesource.com/c/src/+/90251

Bug: chromium:690537
Change-Id: Ieb0eb8d5487e03fc55a221f10366ed9768a6eb16
Reviewed-on: https://webrtc-review.googlesource.com/95061
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24385}
2018-08-22 10:39:28 +00:00
50635036db Add missing ifdefs to header files for SW video codecs.
Some functions were removed from the implementation files when this
flag is unset, but remained in the header files.

Bug: webrtc:7925
Change-Id: I9f8f969fb52f83c05ba02500a62dee78d1bcb0dc
Reviewed-on: https://webrtc-review.googlesource.com/80260
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24366}
2018-08-21 16:20:23 +00:00
58b228461d Simulcast screenshare adjustment to temporal layers, bitrate
Change experimental max bitrate setting from 1.6Mbps to 1.25Mbps in
order to allow a larger fraction of participants to receive this layer.

Add a new field trial to allow setting the number of temporal layers for
the high-quality simulcast stream in screensharing separately from the
temporal layer count for regular video.

Bug: webrtc:9477
Change-Id: I1341b774f870c50710901da24963bd3ede96ffd8
Reviewed-on: https://webrtc-review.googlesource.com/95101
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24356}
2018-08-21 11:48:00 +00:00
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.

Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
2018-08-21 09:15:51 +00:00
820ebd0f66 Add field trial flag for increased receive buffers
Bug: webrtc:9637
Change-Id: Id84c78fa17fbd959af3ab81209e0636317f3da4b
Reviewed-on: https://webrtc-review.googlesource.com/94768
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24349}
2018-08-20 15:51:16 +00:00
77c8e65b88 Update multiplex encoder to support having augmenting data attached to the video
Multiplex encoder is now supporting attaching user-defined data to the video
frame. This data will be sent with the video frame and thus is guaranteed to
be synchronized. This is useful in cases where the data and video frame need
to by synchronized such as sending information about 3D objects or camera
tracking information with the video stream

Multiplex Encoder with data is implemented in a modular way. A new
VideoFrameBuffer type is created with the encoder. AugmentedVideoFrameBuffer
holds the video frame and the data. MultiplexVideoEncoder encodes both
the frame and data.

Change-Id: I23263f70d111f6f1783c070edec70bd11ebb9868
Bug: webrtc:9632
Reviewed-on: https://webrtc-review.googlesource.com/92642
Commit-Queue: Tarek Hefny <tarekh@google.com>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24297}
2018-08-15 21:54:17 +00:00
948b7e3755 Revert "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.

Reason for revert: Breaks downstream project.

Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
> 
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
> 
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}

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

Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
2018-08-14 07:25:23 +00:00
ced5cfdb35 Add initial support for RtpEncodingParameters max_framerate.
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.

Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
2018-08-13 09:59:04 +00:00
f5f5373372 Delete unused member MediaSenderInfo::packets_cached.
Bug: None
Change-Id: I06e1a2010cc0af4b8a4ea726078fea6b67fa84d5
Reviewed-on: https://webrtc-review.googlesource.com/93281
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24263}
2018-08-10 13:13:54 +00:00
b6b1cacd09 Experimental improvements for simulcast screenshare
* Make shorter 4-frame pattern default if 2 temporal layers are used.
* Make DefaultTemporalLayers usable by upper simulcast stream with 2tl.
* If experimental settings are enable, bump the max bitrate for the top
  stream. Since we're now using probing everywhere the rampup should be
  less of an issue.
* Additionally, fixes an issue in full stack tests, where
  ScopedFieldTrials in an experiment would override the
  --force_fieldtrials specified at command line. Some trials added by
  the test bots caused timeouts without this.

Bug: webrtc:9477
Change-Id: I42410605d416b51c4fbfe5b6b850997484af583c
Reviewed-on: https://webrtc-review.googlesource.com/92883
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24252}
2018-08-09 15:10:55 +00:00
8d2995b865 SimulcastEncoderAdapter should not update maxQp for screencast
Bug: webrtc:9608
Change-Id: I70f10c77df6579a24678842a9d9e7a2a528b0c40
Reviewed-on: https://webrtc-review.googlesource.com/93287
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24241}
2018-08-09 10:21:14 +00:00
d528ad542e Make internal video decoder factory more resilient to incorrect usage
If SW H264 is not supported and a client tries to create such a
decoder from InternalDecoderFactory, we currently crash. This CL
changes so that we log an error and return null from CreateDecoder()
instead.

Bug: webrtc:7925
Change-Id: I0c495f62dae25ac0bf4931c02527eb9977db3d92
Reviewed-on: https://webrtc-review.googlesource.com/92395
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24220}
2018-08-08 09:06:26 +00:00
7008287219 Delete struct webrtc::PacketTime.
Replaced by a int64_t representing time in us.

Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
2018-08-07 10:07:15 +00:00
731a2c2dc6 Convert webrtcvideoengine CVO tests away from cricket::VideoCapturer.
Bug: webrtc:6353
Change-Id: I1f4f705cda4fdf88465395898e2588b2a19eebf3
Reviewed-on: https://webrtc-review.googlesource.com/83324
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24142}
2018-07-30 14:41:23 +00:00
17aff35e1d Enable clang::find_bad_constructs for sdk/ (part 1).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I6f03c46e772ccf4d15951a4b9d4e12015d539e58
Reviewed-on: https://webrtc-review.googlesource.com/90408
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24113}
2018-07-26 12:16:31 +00:00
a3df0f2d05 Remove simulcast screenshare experimental code
Bug: chromium:690537
Change-Id: I2ed850eb7e450e9666aeb7cc3b55db073ed5a8a9
Reviewed-on: https://webrtc-review.googlesource.com/90251
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24104}
2018-07-25 16:32:34 +00:00
e41c433502 Move sigslot to proper third_party directory
Extract sigslot into separate target and move it to proper third_party
directory.

Bug: webrtc:8366
Change-Id: Id2e0712bd020bfad811947803c94553dce06d976
Reviewed-on: https://webrtc-review.googlesource.com/84141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 14:53:33 +00:00
e507b0ce8e Turn off comfort noise generation by default in AECM
All clients who do not own their own APM turn it off by default
(in WebrtcVoiceEngine). AECM with comfort noise is a little-exercised
code path. Configurability of this setting is going away, so we're
better off disabling it by default.

Bug: webrtc:9535
Change-Id: Iba839aa18e79ae29ff20bdf6e30de77870ba4143
Reviewed-on: https://webrtc-review.googlesource.com/89583
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24078}
2018-07-24 08:52:36 +00:00
a76af0ca2e Move base64.h to the proper location.
Move base64.h to the proper location and put redirect header into the
old place to be able to switch downstream users on new location.

Bug: webrtc:8366
Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291
Reviewed-on: https://webrtc-review.googlesource.com/88223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 15:40:36 +00:00
3d72f6dac8 Fixing -Wsign-compare errors.
Error example:
absl/types/optional.h:1107:54: error: comparison of integers of
different signs: 'const unsigned long' and 'const int'
[-Werror,-Wsign-compare]
[...]
EXPECT_GT(streams[0].num_temporal_layers, 1);

Bug: None
Change-Id: Ifa84e318e242d0dfb32a4f2166464d91fcc86fb1
Reviewed-on: https://webrtc-review.googlesource.com/89744
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24060}
2018-07-23 08:06:09 +00:00
79eb4dd928 Enabling clang::find_bad_constructs for libjingle_peerconnection_api.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I5475e574353c772910181495fdb3400b5f0e7399
Reviewed-on: https://webrtc-review.googlesource.com/87240
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24040}
2018-07-19 09:17:10 +00:00
4f6d233dcc Added explicit EOR to sctp messages and coalesce messages on the receiving side.
TBR=pthatcher@webrtc.org

Bug: webrtc:7774
Change-Id: I41d1cd98d1e7b2ad479177eb2e328a5e2c704824
Reviewed-on: https://webrtc-review.googlesource.com/88900
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24031}
2018-07-19 01:26:59 +00:00
87b3c510b4 Implement changing degradation preference with setParameters()
The current default behavior is unchanged and points to MAINTAIN_FRAMERATE,
meaning there is no way to currently use BALANCED as we can't detect
when the value as been set or not.
Updating this is an API change that should be done in another CL and
properly communicated first.


Bug: webrtc:7607
Change-Id: Ic3877ad8dd7bc418296f21a04bc37f59ec55934a
Reviewed-on: https://webrtc-review.googlesource.com/88766
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24024}
2018-07-18 14:45:27 +00:00
d93a51dfaa Enabling clang::find_bad_constructs for common_video.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I6d3b45de9dca3a5a04f0cdd5583919d35a585a7e
Reviewed-on: https://webrtc-review.googlesource.com/89043
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24018}
2018-07-18 11:26:01 +00:00
89b2963810 Reland "Enable simulcast screenshare by default"
This is a reland of d43c692ba7f53b5576a494c0343bc7a4bb36831b after fixes
to failing chromium tests. No change to the original CL were done.
Original CL reviewed on: https://webrtc-review.googlesource.com/87560

TBR=stefan@webrtc.org

Bug: chromium:690537
Change-Id: I6b59ffc90d789aff21c7e52b118d3dfbe756c8a9
Reviewed-on: https://webrtc-review.googlesource.com/89081
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24013}
2018-07-18 08:58:09 +00:00
ca536d4692 Revert "Enable simulcast screenshare by default"
This reverts commit d43c692ba7f53b5576a494c0343bc7a4bb36831b.

Reason for revert: Breaks chromium unit tests

Original change's description:
> Enable simulcast screenshare by default
> 
> Bug: chromium:690537
> Change-Id: I8b713a9c4d9d5d1a5cf13dff607cc25806aceed2
> Reviewed-on: https://webrtc-review.googlesource.com/87560
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24003}

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

Change-Id: I55b952519458bb9ab49cf6377601d7420e71d086
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:690537
Reviewed-on: https://webrtc-review.googlesource.com/89080
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24005}
2018-07-17 12:47:55 +00:00
d43c692ba7 Enable simulcast screenshare by default
Bug: chromium:690537
Change-Id: I8b713a9c4d9d5d1a5cf13dff607cc25806aceed2
Reviewed-on: https://webrtc-review.googlesource.com/87560
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24003}
2018-07-17 11:22:37 +00:00