Commit Graph

32 Commits

Author SHA1 Message Date
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
00c7183614 Replace rtc::Optional with absl::optional in media, ortc, p2p
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'media ortc p2p':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I19167714af7cc1436d34cfcba6c8b3718d8e677b
Reviewed-on: https://webrtc-review.googlesource.com/83731
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23638}
2018-06-16 07:09:59 +00:00
abe301fe6c Add HeaderExtensions to RtpParameters
Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}
2018-06-12 17:01:40 +00:00
0327c2ddc1 Move VideoStreamEncoderInterface to api/.
Bug: webrtc:8830
Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0
Reviewed-on: https://webrtc-review.googlesource.com/74481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 19:50:37 +00:00
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
49fcc10de6 Merge DegradationPreference enums.
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".

It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.

Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
2018-05-17 11:21:52 +00:00
c6ce9c5938 New file api/video/BUILD.gn
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.

Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
4db138e889 Reland "Move creating encoder to VideoStreamEncoder."
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80

Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}

TBR=magjed@webrtc.org,kwiberg@webrtc.org

Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
2018-04-19 08:48:58 +00:00
0d650b44ef Revert "Move creating encoder to VideoStreamEncoder."
This reverts commit fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80.

Reason for revert: Appears to break Chromium, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Linux%20Tester/43756, where remoting_unittests failed.

Original change's description:
> Move creating encoder to VideoStreamEncoder.
> 
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
> 
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}

TBR=magjed@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: I47ee3ac42e62472d825a08c98e28f9ae53ec9fff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/70600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22914}
2018-04-18 07:17:16 +00:00
fb82fcc7f9 Move creating encoder to VideoStreamEncoder.
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.

Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
2018-04-17 15:04:33 +00:00
ff40b142c0 Delete obsolete enable argument to SetVideoSend.
This argument was previously used to implement track muting
(black frames) in the video engine, but that now happens in
the VideoTrack/VideoBroadcaster upstream.

Bug: webrtc:6983
Change-Id: Ib721b297d9fbe55b641c56690dbbd37a52edbb2f
Reviewed-on: https://webrtc-review.googlesource.com/67341
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22785}
2018-04-09 08:45:29 +00:00
5897a6ec6a Adds support for signaling a=msid lines without a=ssrc lines.
Currently in the SDP we require an a=ssrc line in the m= section in
order for a StreamParams object to be created with that
MediaContentDescription. This change creates a StreamParams object
without ssrcs in the case that a=msid lines are signaled, but ssrcs
are not. When the remote description is set, this allows us to store
the "unsignaled" StreamParams object in the media channel to later
be used when the first packet is received and we create the
receive stream.

Bug: webrtc:7932, webrtc:7933
Change-Id: Ib6734abeee62b8ed688a8208722c402134c074ef
Reviewed-on: https://webrtc-review.googlesource.com/63141
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22712}
2018-04-03 21:21:11 +00:00
88614b0b06 Pass VideoEncoderFactory from WebrtcVideoEngine to VideoStreamEncoder.
In preparation for also moving the responsibility for encoder creation.

Bug: webrtc:8830
Change-Id: Ic3d2039a86cd31c1b4157f5df4e97b607c81f1d7
Reviewed-on: https://webrtc-review.googlesource.com/55264
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22630}
2018-03-27 15:52:01 +00:00
bb50ce5bb6 Wire up MID send value to the PeerConnection API
Bug: webrtc:4050
Change-Id: I522cf8621e2cb639f54be2402174befd23e4af59
Reviewed-on: https://webrtc-review.googlesource.com/60962
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22610}
2018-03-26 18:14:30 +00:00
5bf8ccdfa0 Delete encoder caching in WebRtcVideoSendStream.
This is a followup to https://webrtc-review.googlesource.com/61640,
which ensures that picture id and tl0 pic idx are continuous,
independent of how the encoder objects are created and destroyed.

The plan is to later move responsibility for encoder creation to
VideoSendStream::ReconfigureVideoEncoder, delegating work to
VideoStreamEncoder.

Bug: webrtc:8830
Change-Id: Idde5c91f24d3c0e3fa6a3bb26eb06f6800896a28
Reviewed-on: https://webrtc-review.googlesource.com/62082
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22473}
2018-03-16 13:20:21 +00:00
fc8d26bd8a Reland "Moved BitrateConfig out of Call::Config."
This is a reland of 5897fe27abcbe70f706cc23adc26147e0581f97e.

Adding back CallConfig::kDefaultStartBitrateBps as deprecated.
Also making BitrateContraints::kDefaultStartBitrateBps private to stop
it from being used in other places.

Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}

Bug: webrtc:8415
Change-Id: Iacfe2d6daedff710832ab89210c7c66d4403c93b
Reviewed-on: https://webrtc-review.googlesource.com/55980
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22123}
2018-02-21 11:38:42 +00:00
e4bf600cad Revert "Moved BitrateConfig out of Call::Config."
This reverts commit 5897fe27abcbe70f706cc23adc26147e0581f97e.

Reason for revert: Breaking internal builds

Original change's description:
> Moved BitrateConfig out of Call::Config.
> 
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
> 
> Also renaming BitrateConfig to BitrateConstraints.
> 
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}

TBR=nisse@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: I598040edba7f1ff8b39d2d9c3c3ceca5627aaa0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/55740
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22106}
2018-02-20 19:16:38 +00:00
5897fe27ab Moved BitrateConfig out of Call::Config.
This prepares for a CL extracting the bitrate configuration logic from
the Call class.

Also renaming BitrateConfig to BitrateConstraints.

Bug: webrtc:8415
Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
Reviewed-on: https://webrtc-review.googlesource.com/54400
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22104}
2018-02-20 16:40:05 +00:00
1370e309e7 Refactor of GetSimulcastConfig & EncoderStreamFactory.
The main pieces of this refactor are splitting up the creation of
simulcast layers for screenshare or the normal case, more consistent
naming, renaming streams to layers and trying to be more explicit with
some of the logic. Also added TODOs for future work to put more
application control into creating simulcast streams.

Bug: webrtc:8785
Change-Id: Ibf49fa0cc6d890ff96f8ee11c89d93a2c94119d6
Reviewed-on: https://webrtc-review.googlesource.com/47580
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21989}
2018-02-12 20:08:34 +00:00
1e06289cdb Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.

Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07 10:07:28 +00:00
dd8c16574e Enable building WebRTC without built-in software codecs
This CL adds a GN build flag to include builtin software codecs
(enabled by default).

When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.

Replaces https://webrtc-review.googlesource.com/c/src/+/29203

Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-31 08:33:59 +00:00
ba37b4b075 Change return type of RtpSenderInterface::SetParameters from bool to RTCError
Note that RTCErrorTypes are mapped to the following DOMException names:
INTERNAL_ERROR -> OperationError
UNSUPPORTED_PARAMETER -> OperationError
INVALID_STATE -> InvalidStateError
INVALID_MODIFICATION -> InvalidModificationError
INVALID_RANGE -> RangeError

Bug: webrtc:8772
Change-Id: I44e3fe2456b007b8fb227d37d74b07ba226a19e4
Reviewed-on: https://webrtc-review.googlesource.com/37141
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21766}
2018-01-25 19:06:04 +00:00
9e19403d10 Move videosourceinterface to api.
VideoSourceInterface is clearly an integral part of
mediastreaminterface.h already, so moving that interface to api makes
sense. This also resolves a circular dependency in call/.

Bug: webrc:6828
Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c
Reviewed-on: https://webrtc-review.googlesource.com/37500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21498}
2018-01-05 09:14:19 +00:00
be214a26f8 Move videosinkinterface.h to common_video to solve a circular dep.
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.

Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
5f5918f4ef Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks.
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.

This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520

Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20678}
2017-11-14 20:42:36 +00:00
fb65b10fa1 Removing forward declarations that are never referenced.
Bug: None
Change-Id: I0f472f55671e92d824c672f3c3b1bc083b8440fc
Reviewed-on: https://webrtc-review.googlesource.com/22004
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20631}
2017-11-10 13:49:45 +00:00
07e0d015b7 Reland "Add helper functions for migrating to new video codec factories"
This reverts commit a7678667fc726979ca27d2429689d5735cca425d.

Reason for reland: Fix initializer list constructor.

Original change's description:
> Revert "Add helper functions for migrating to new video codec factories"
> 
> This reverts commit 1c9623c70db42550d152c127e01434004087f743.
> 
> Reason for revert: Breaks brace initialization:
> 
> cricket::VideoDecoderParams params = { "deadbeef" };
> 
> I suggest adding an initializer list constructor.
> 
> Original change's description:
> > Add helper functions for migrating to new video codec factories
> > 
> > This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> > convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> > webrtc::VideoEncoder and webrtc::VideoDecoder.
> > 
> > The purpose is to make it as easy as possible for clients to migrate to the new
> > API and allow us to stop depending on the internal SW codecs as soon as possible.
> > 
> > There still exists an ugly decoder adapter class in the video engine. The reason
> > is that we need to continue to pass in the |receive_stream_id| decoder params to
> > some legacy clients.
> > 
> > Bug: webrtc:7925
> > Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> > Reviewed-on: https://webrtc-review.googlesource.com/15181
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20475}
> 
> TBR=magjed@webrtc.org,andersc@webrtc.org
> 
> Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7925
> Reviewed-on: https://webrtc-review.googlesource.com/17160
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20486}

TBR=deadbeef@webrtc.org,magjed@webrtc.org,andersc@webrtc.org

Change-Id: Ic825d133b6e1c6e5aad811ba528751dd5ed85e67
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20501}
2017-10-31 13:32:22 +00:00
a7678667fc Revert "Add helper functions for migrating to new video codec factories"
This reverts commit 1c9623c70db42550d152c127e01434004087f743.

Reason for revert: Breaks brace initialization:

cricket::VideoDecoderParams params = { "deadbeef" };

I suggest adding an initializer list constructor.

Original change's description:
> Add helper functions for migrating to new video codec factories
> 
> This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
> convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
> webrtc::VideoEncoder and webrtc::VideoDecoder.
> 
> The purpose is to make it as easy as possible for clients to migrate to the new
> API and allow us to stop depending on the internal SW codecs as soon as possible.
> 
> There still exists an ugly decoder adapter class in the video engine. The reason
> is that we need to continue to pass in the |receive_stream_id| decoder params to
> some legacy clients.
> 
> Bug: webrtc:7925
> Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
> Reviewed-on: https://webrtc-review.googlesource.com/15181
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20475}

TBR=magjed@webrtc.org,andersc@webrtc.org

Change-Id: I0d1084dc86979fbca748d9ba287d1db3dbe52b44
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/17160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20486}
2017-10-30 22:52:59 +00:00
1c9623c70d Add helper functions for migrating to new video codec factories
This CL adds helper functions in media/engine/convert_legacy_video_factory.h to
convert from the old WebRtcVideoEncoder and WebRtcVideoDecoder to the new
webrtc::VideoEncoder and webrtc::VideoDecoder.

The purpose is to make it as easy as possible for clients to migrate to the new
API and allow us to stop depending on the internal SW codecs as soon as possible.

There still exists an ugly decoder adapter class in the video engine. The reason
is that we need to continue to pass in the |receive_stream_id| decoder params to
some legacy clients.

Bug: webrtc:7925
Change-Id: I43ff03e036411a85d4940fe517a34489f171d698
Reviewed-on: https://webrtc-review.googlesource.com/15181
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20475}
2017-10-30 14:18:30 +00:00
02e7a1981a Remove unnecessary video factory references in PeerConnectionFactory
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.

Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
2017-09-27 14:41:46 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00