This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.
Reason for revert: Reland by removing the conflict with the broken CL.
Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
>
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
>
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
>
> This causes a merge conflict. So need to revert this first.
>
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> >
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> >
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> >
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
>
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org
Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
This causes a merge conflict. So need to revert this first.
Original change's description:
> Move allocation and rtp conversion logic out of payload router.
>
> Makes it easier to write tests, and allows for moving rtp module
> ownership into the payload router in the future.
>
> The RtpPayloadParams class is split into declaration and definition and
> moved into separate files.
>
> Bug: webrtc:9517
> Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> Reviewed-on: https://webrtc-review.googlesource.com/88564
> Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23983}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/88821
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23991}
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.
The RtpPayloadParams class is split into declaration and definition and
moved into separate files.
Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
This is an reland of 6f5b0f920af08d66e6b77ee4f91ade5797145368
Relanded after speculative revert without any changes.
TBR=ilnik@webrtc.org
Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}
Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.
Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
This CL adds a helper class GlShaderBuilder to build an instances of
RendererCommon.GlDrawer that can accept multiple input sources
(OES, RGB, or YUV) using a generic fragment shader as input.
Bug: webrtc:9355
Change-Id: I14a0a280d2b6f838984f7b60897cc0c58e2a948a
Reviewed-on: https://webrtc-review.googlesource.com/80940
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23622}
This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd
Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.
Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}
Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
Previously, constructing a PeerConnection or WebRtcVideoEngine with
fake encoder/decoder factories would result in the real, built-in factories
also being used. In https://webrtc-review.googlesource.com/c/src/+/71162, this
changed, so to temporarily allow tests to continue working exactly the same as
before, the fake factories started encapsulating the real factories. This CL
removes that behavior and updates the tests accordingly.
Bug: webrtc:9228
Change-Id: Ida14a1e3f5f5a0e2f03100b7895b3b1bdf0a0a42
Reviewed-on: https://webrtc-review.googlesource.com/75260
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23209}
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}
The goal is to make these injectable, and only VP8 and VP9 specific
targets should depend on them.
Bug: webrtc:7925
Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2
Reviewed-on: https://webrtc-review.googlesource.com/72082
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23021}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL adds a field trial to enable the BBR congestion control method.
Since BBR is only implemented to handle per packet feedback,
SendSideCongestionController is modified to recreate network controllers
when the packet feedback availability changes and the BBR experiment is
enabled.
This also means that the periodic task used for process updates in the
network controllers has to recreated.
Bug: webrtc:8415
Change-Id: Ia24f7ad35336d2cc7a02bb3a445f1a84b8643475
Reviewed-on: https://webrtc-review.googlesource.com/61520
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@{#22791}
|is_posix| will be switched to false for Fuchsia, this is a preliminary change.
Bug: chromium:812974
Change-Id: I3bfda3e056ad1e5229834286ce5d095d9204a428
Reviewed-on: https://webrtc-review.googlesource.com/65782
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22753}
Implements JavaToNativeStringMap that is a replacement for
JavaToStdMapStrings. It uses a new template method JavaToNativeMap. Also
adds testing support for native API and a test for JavaToNativeStringMap.
Bug: webrtc:8769
Change-Id: I580d4992a899ebe02da39af450fa51d52ee9b88b
Reviewed-on: https://webrtc-review.googlesource.com/48060
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21967}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
1. Added WEBRTC_FUCHSIA define.
2. Added PlatformThreadId typedef for Fuchsia.
3. Updated ifdefs for _strnicmp()/strncasecmd(), so _strnicmp()
is used on all platforms
3. Updated ifdefs in clock.cc to avoid invalid assumption that
POSIX = LINUX || MAC .
Bug: chromium:750940
Change-Id: Id7aa98e017f467bcebb78a0b298ba91655502072
Reviewed-on: https://webrtc-review.googlesource.com/31641
Commit-Queue: Sergey Ulanov <sergeyu@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21233}
If disabled, this allows running "gn gen <folder>" on a system without
GTK and other development libraries installed.
BUG=webrtc:5716
Change-Id: I1aee3d74402485113612276f6de374fb6034251f
Reviewed-on: https://webrtc-review.googlesource.com/30460
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21192}
To be honest I don't know why the common_types.h includes from rtc_base
don't trip the check. I'd expect that to require that webrtc_common
depends on rtc_base, and in that case we're in trouble.
Bug: webrtc:6828,webrtc:7745
Change-Id: Id15055483e4fe40cffa67b973cc54cae3ed26112
Reviewed-on: https://webrtc-review.googlesource.com/28041
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21156}
This target is just confusing, since it runs essentially random tests
out of the test binaries it's depending on. This CL makes the top
GN file structured like Chromium's is.
A possible secondary function is to act as a link test (maybe to catch
twice-defined symbols etc), but that seems highly speculative.
Bug: None
Change-Id: Ic8a36ffe96408312e602a58f8806b0694fedaad6
Reviewed-on: https://webrtc-review.googlesource.com/27380
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21028}
