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}
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}
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}
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}
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}
This makes the sequence of expected calls easier to read. Also, we can
save one line and get rid of a gmock warning by expecting the
MockAudioEncoder object to be destroyed at the end of the test instead
of making a final marker call.
Review URL: https://codereview.webrtc.org/1331793003
Cr-Commit-Position: refs/heads/master@{#9916}
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.
BUG=chromium:468375 (in particular comment #37)
NOTRY=true
Review URL: https://codereview.webrtc.org/1316363005
Cr-Commit-Position: refs/heads/master@{#9913}
Fix bug 4981, which caused the second half (decreasing loss rates) to
not test anything. In the process, the test is changed slightly to
make it less dependent on the exact rounding behavior of doubles (by
not testing exactly at the the points where the effective loss rate
goes through a step---just very very close). A bunch of symbolic
constants are also replaced with easy-to-read literal numbers.
BUG=4981
Review URL: https://codereview.webrtc.org/1316673010
Cr-Commit-Position: refs/heads/master@{#9908}
Merge the contents of audio_encoder_mutable_opus_test.cc into
audio_encoder_opus_unittest.cc, since they're now both testing
AudioEncoderOpus.
(While preparing this CL, I noted a bug in the PacketLossRateOptimized
test. This CL leaves that test essentially unchanged; I've posted bug
4981 about the problem.)
Review URL: https://codereview.webrtc.org/1319713004
Cr-Commit-Position: refs/heads/master@{#9906}
And the corresponding ACM methods SetISACMaxRate and
SetISACMaxPayloadSize. They were only used in tests.
Review URL: https://codereview.webrtc.org/1311533010
Cr-Commit-Position: refs/heads/master@{#9903}
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.
Review URL: https://codereview.webrtc.org/1332573003
Cr-Commit-Position: refs/heads/master@{#9900}
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.
Review URL: https://codereview.webrtc.org/1322973004
Cr-Commit-Position: refs/heads/master@{#9894}
Unwrap uint16_t to int64_t, based on delta and last sequence number.
This can make application logic, putting packets in maps etc, much
simpler.
BUG=
Review URL: https://codereview.webrtc.org/1209623002
Cr-Commit-Position: refs/heads/master@{#9887}
This CL adds support for simulating neteq using stored RTP packets as well as calls to GetAudio from an RtcEventLog, using the stored timestamps.
The type of the input file is detected automatically.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1316903002
Cr-Commit-Position: refs/heads/master@{#9886}
Handling the case when encoder drops only the higher layer.
Added options to screenshare loopback test to discard high temporal or spatial layers (to view the lower layers).
Review URL: https://codereview.webrtc.org/1287643002
Cr-Commit-Position: refs/heads/master@{#9883}
Part of work removing dependency on Chromium's base.
BUG=468375 (in particular comment #37)
Review URL: https://codereview.webrtc.org/1327023002
Cr-Commit-Position: refs/heads/master@{#9880}
