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}
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}
These were accidentally disabled due to checking ssrcs_.size() (which
includes RTX SSRCs) instead of rtp.ssrcs.size() to determine whether a
stream is simulcast or not.
BUG=webrtc:4965
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1318193003 .
Cr-Commit-Position: refs/heads/master@{#9907}
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}
