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}
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}
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}
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}
* VoEBase contains only stub methods (until downstream code is
updated).
* voe::Channel and ChannelProxy classes remain, but are now created
internally to the streams. As a result,
internal::Audio[Receive|Send]Stream can have a ChannelProxy injected
for testing.
* Stream classes share Call::module_process_thread_ for their RtpRtcp
modules, rather than using a separate thread shared only among audio
streams.
* voe::Channel instances use Call::worker_queue_ for encoding packets,
rather than having a separate queue for audio (send) streams.
Bug: webrtc:4690
Change-Id: I8059ef224ad13aa0a6ded2cafc52599c7f64d68d
Reviewed-on: https://webrtc-review.googlesource.com/34640
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21578}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:6000, webrtc:8396
Change-Id: Ibb6b3f36f14b956c55d4edc934d101cb855b272d
Reviewed-on: https://webrtc-review.googlesource.com/18420
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20622}
This CL exposes the new type of video codec factories that represent all
video codecs in the PeerConnectionFactory API, i.e. no extra internal SW
video codecs will be added. Clients of the new functions will be
responsible for adding all SW video codecs themselves, and also handling
SW fallback and simulcast.
BUG=webrtc:7925
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/3004353002 .
Cr-Commit-Position: refs/heads/master@{#19866}
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}
