All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1346993002
Cr-Commit-Position: refs/heads/master@{#9966}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
The WebKit dependency could be removed again after
https://codereview.chromium.org/1338193003/
Relevant changes:
* src/buildtools: 565d04e..f7310ee
* src/third_party/boringssl/src: ac8302a..1d128f3
* src/third_party/libvpx: 0304cef..ac1772e
* src/third_party/libyuv: 0bc626a..fcacbfb
* src/third_party/mockito/src: ed99a52..4d987dc
* src/tools/swarming_client: 2866a22..77f720b
Details: 5482f56..310ea93/DEPS
Clang version was not updated in this roll.
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1347153003 .
Cr-Commit-Position: refs/heads/master@{#9962}
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1348053002
Cr-Commit-Position: refs/heads/master@{#9961}
Recent changes (https://codereview.chromium.org/1311013010) introduces a
dependency on WebKit (Blink) in Chromium, which forces us to start pulling
down that as well (+6GB). However Blink is about to be merged into the
Chromium repo soon anyway, so the size increase is inevitable.
Luckily, this can be removed in the next roll, if we roll past
http://crrev.com/348812
The ijar dependency was introduced in https://codereview.chromium.org/1323053003 (#347208)
Relevant changes:
* src/third_party/boringssl/src: 12fe1b2..ac8302a
* src/third_party/libvpx: a208eca..0304cef
* src/third_party/libyuv: 3c4f573..0bc626a
* src/tools/gyp: 6ee91ad..5d01a8c
Details: a28d8d5..5482f56/DEPS
Clang version was not updated in this roll.
R=torbjorng@webrtc.orgTBR=marpan@webrtc.org
BUG=webrtc:5005, chromium:530112
Review URL: https://codereview.webrtc.org/1305043008 .
Cr-Commit-Position: refs/heads/master@{#9956}
Part of work removing dependency on Chromium's base.
Only adds "= delete". From https://codereview.chromium.org/1151443003 :
"This will guarantee the error to be at compile time, and not rely on the call visibility (private)."
In consequence of that change, fixed an illegal copy and removed a bunch of unused variables.
Depends on https://codereview.webrtc.org/1345433002/
BUG=chromium:468375
(in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1342543004
Cr-Commit-Position: refs/heads/master@{#9954}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
When GYP runs for OS=android it doesn't generate the
video_engine_core_unittests_apk_target target which is needed to
get the APK built.
The same problem applies to webrtc/test/webrtc_test_common.gyp,
but that unittest is not added on any bot anyway, so that's future work.
TESTED=Ran webrtc/build/gyp_webrtc for Linux and Android locally.
Before this patch, the video_engine_core_unittests was not built
as part of the 'All' target. With this patch, it is now built.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1348093002 .
Cr-Commit-Position: refs/heads/master@{#9952}
This CL should not do any functional changes. It removes some redundant arguments and unnecessary error checking.
BUG=webrtc:4993
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1338943003 .
Cr-Commit-Position: refs/heads/master@{#9950}
Reason for revert:
Breaks goma (??!??!?)
Original issue's description:
> Bailing out if pc factory fails to get created.
>
> This prevents us from continuing if we fail initialization.
> The failure will happen closer to its source, rather than
> when we try to create the first peer connection.
>
> BUG=None
> R=glaznev@webrtc.org
>
> Committed: https://crrev.com/6eb75d9e67f02c256436eb96f3c77026486561a1
> Cr-Commit-Position: refs/heads/master@{#9948}
TBR=glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review URL: https://codereview.webrtc.org/1344363002
Cr-Commit-Position: refs/heads/master@{#9949}
This prevents us from continuing if we fail initialization.
The failure will happen closer to its source, rather than
when we try to create the first peer connection.
BUG=None
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1339923004 .
Cr-Commit-Position: refs/heads/master@{#9948}
Need to figure out the best way to initialize native logging system
while peer connection factory is not created yet.
R=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1343163003 .
Cr-Commit-Position: refs/heads/master@{#9947}
Currently disposing Java peer connection object will result in auto
release of media streams and media tracks added to peer connection.
Add an option to allow application to own video track so it can be
kept after peer connection is destroyed.
R=jiayl@webrtc.org, wzh@webrtc.org
Review URL: https://codereview.webrtc.org/1333363002 .
Cr-Commit-Position: refs/heads/master@{#9946}
Remove start_bitrate_bps which is no longer used and return the current
allocated bitrate instead of having it as an out parameter, removing the
previous return value which is no longer used.
Permits removing bitrate controller usage from ViEEncoder.
BUG=webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1343783006 .
Cr-Commit-Position: refs/heads/master@{#9942}
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1269863005 .
Cr-Commit-Position: refs/heads/master@{#9939}
Add new helper class to create and synchronize access to SurfaceTextures. The plan is replace the SurfaceTexture in MediaCodecVideoDecoder in a follow-up CL and remove the SurfaceTexture.updateTexImage() call in VideoRendererGui.
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1342713003 .
Cr-Commit-Position: refs/heads/master@{#9938}
Future log messages should all be sent to org.webrtc.Logging as well.
BUG=
Review URL: https://codereview.webrtc.org/1338033003
Cr-Commit-Position: refs/heads/master@{#9936}
Incoming frames usually have an epoch of time since the capturer was
created or similar, not any fixed-time epoch. As such, setting a new
capturer resulted in delivering frames with older timestamps which
caused these frames to be dropped before encoding.
BUG=webrtc:4994
R=stefan@webrtc.orgTBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345473002
Cr-Commit-Position: refs/heads/master@{#9934}
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
Helps differentiate between different instances when debugging.
Review URL: https://codereview.webrtc.org/1337003003
Cr-Commit-Position: refs/heads/master@{#9927}
Collects packet information within a struct instead of spreading it out
over different vectors. Adds a fixed-size buffer to the stored packet
instead of using vectors.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1340573002
Cr-Commit-Position: refs/heads/master@{#9926}
It is possible that cameraThreadHandler is null upon calling
switchCamera(). This CL adds a guard against that.
Review URL: https://codereview.webrtc.org/1325333003
Cr-Commit-Position: refs/heads/master@{#9925}
This CL ensures that we return -1 in cases where InitRecording() fails. It ensures that we don't crash applications.
BUG=b/22849644
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1323243012 .
Cr-Commit-Position: refs/heads/master@{#9918}
If we don't, we'll end up crashing if they're enabled when the speech
encoder is in the middle of encoding a packet, since CNG and RED
assume that the speech encoder starts out with an empty buffer
(because they need to be in sync with it).
BUG=chromium:490368
Review URL: https://codereview.webrtc.org/1331853002
Cr-Commit-Position: refs/heads/master@{#9917}
