Test enables single-nalu mode, sets limit for nalu lenght and verifies
that encoder follows that limit.
I found that QP jumps significantly when the mode is enabled. In result
encoder might produce 4kbyte and 0.4kbyte frames back-to-back. But it
seems that happens only to couple of frames in the beginning. This
caused test to fail with default RC thresholds. To bypass this I
increased frame size mismatch threshold from 20 to 30%. This should be
Ok considering single-nalu mode is rare.
BUG=webrtc:8070
Review-Url: https://codereview.webrtc.org/3014623002
Cr-Commit-Position: refs/heads/master@{#20023}
Also include a small fix for getting the encoder queue.
Bug: webrtc:7760
Change-Id: I96dc8ffb363b90382276d88148f81d5f89dca5f2
Reviewed-on: https://webrtc-review.googlesource.com/2683
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20022}
If this is not done, the RTC_DCHECK_CALLED_SEQUENTIALLY might fire
if the encoder is used on a new VideoStreamEncoder. This happens
after VideoSendStream recreations due to changes in the SDP.
BUG=b/66590444
Change-Id: I086370526afbbe2ba629805f97f89e512ba3f472
Reviewed-on: https://webrtc-review.googlesource.com/4360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20020}
There is bug in AQ in v1.6.0 which causes raising QP to maximum value
and results in very poor quality of video no matter of allocated bitrate.
To trigger this someone needs to set packetization_mode=0 (or just do
not transmit this flag at all) in SDP. In this mode the encoder
enables multi-slice and disables AQ partially such that some part of
AQ algo still works and leads QP to maximum. The issue is fixed in v1.7.0.
BUG=webrtc:8070
Review-Url: https://codereview.webrtc.org/3011373002
Cr-Commit-Position: refs/heads/master@{#20015}
ReportBlock is the the real receiver report.
Triggering rtt update on ReportBlock support clients that send receiver
report blocks attached to SenderReport rather than ReceiverReport.
Bug: webrtc:7996
Change-Id: Ie826fa09fd1bf0e5256e995649f66811b5192761
Reviewed-on: https://webrtc-review.googlesource.com/4040
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20014}
This is a reland of 7a2bfd22e69f14e2af989b9e30ddd834f585caa9
Original change's description:
> Improve unit testing for HardwareVideoEncoder and fix bugs.
>
> Improves the unit testing for HardwareVideoEncoder and fixes bugs in it.
> The main added feature is support for dynamically switching between
> texture and byte buffer modes.
>
> Bug: webrtc:7760
> Change-Id: Iaffe6b7700047c7d0f9a7b89a6118f6ff932cd9b
> Reviewed-on: https://webrtc-review.googlesource.com/2682
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19963}
Bug: webrtc:7760
Change-Id: I605647da456525de8e535cc66cab9d0b3f14240b
Reviewed-on: https://webrtc-review.googlesource.com/3641
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20013}
This adds four parameters to the protobuf that is used to configure the ANA controllers. These extra parameters allow for setting an offset to the per-packet overhead that is used when changing the frame length size and when changing bitrate.
BUG=webrtc:8179
Review-Url: https://codereview.webrtc.org/3013613002
Cr-Commit-Position: refs/heads/master@{#20011}
If the network interface appears active, but binding the sockets fails,
then it won't produce any candidates even though it's never marked as
"network failed". So this was causing nothing to happen once a network
change event occurs and the interface becomes usable again.
So, this CL adds the condition that we only disable gathering of local
ports if we don't have them already.
See bug for more details.
BUG=webrtc:8256
Review-Url: https://codereview.webrtc.org/3015543002
Cr-Commit-Position: refs/heads/master@{#20007}
This lays the groundwork for splitting up the
PeerConnectionInterface unit tests into multiple files so that
the tests can be organized better. The intent is for each unit
test file to declare a test fixture which subclasses
PeerConnectionUnitTestFixture and creates PeerConnectionWrappers
to write assertions against.
Bug: webrtc:8222
Change-Id: I21175b1e1828a6cd5012305a8a27faaf4eecf81c
Reviewed-on: https://webrtc-review.googlesource.com/1120
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20004}
The video codec factories should be owned by the video engine instead
of by the PeerConnectionFactory.
Bug: None
Change-Id: If63d47cef565138d51377af3fc9ea973950c9390
Reviewed-on: https://webrtc-review.googlesource.com/1601
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20002}
On android, the flag to store the frame with the worst PSNR was called
'--test_artifacts_dir'.
I think test artifacts is a better name.
TBR=sprang@webrtc.org
Bug: chromium:745469
Change-Id: I358ea2985a1df2da12b81df173d74ac193556a49
Reviewed-on: https://webrtc-review.googlesource.com/4080
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20000}
This reverts commit ba78b5a905bffa05933a135673996df02328f2a4.
Reason for revert: Breaks external projects.
Original change's description:
> Android: Generate JNI code for VideoSink and VideoEncoder
>
> This is the first CL to start generating JNI code. It has updated two of
> the most recent classes to use JNI code generation.
>
> Bug: webrtc:8278
> Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
> Reviewed-on: https://webrtc-review.googlesource.com/3820
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19994}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I48e079f3ab9661ae4171a3ae5cca571a75d14810
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278
Reviewed-on: https://webrtc-review.googlesource.com/4100
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19997}
webrtc_perf_tests needs more than 15 min to run.
NOTRY=True
Bug: chromium:755660
Change-Id: Ibabfae3679206105d585c35f80b839f0046f9ccc
Reviewed-on: https://webrtc-review.googlesource.com/4021
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19996}
Tests disabled for Mac only.
Tests fail in this way on perf bot Mac 10.11:
[ RUN ] CallPerfTest.CaptureNtpTimeWithNetworkDelay
../../call/call_perf_tests.cc:407: Failure
Value of: std::abs(time_offset_ms) < threshold_ms_
Actual: false
Expected: true
TBR=stefan@webrtc.org
Bug: webrtc:8291
Change-Id: I8d173fcff21f096827f31ddd670b6647796bff4b
Reviewed-on: https://webrtc-review.googlesource.com/4041
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19995}
This is the first CL to start generating JNI code. It has updated two of
the most recent classes to use JNI code generation.
Bug: webrtc:8278
Change-Id: I1b19ee78c273346ceeaa0401dbdf8696803f16c7
Reviewed-on: https://webrtc-review.googlesource.com/3820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19994}
The gain suggested by AGC is optionally used in audioproc_f to simulate analog gain applied to the mic.
The simulation is done by applying digital gain to the input samples.
This functionality is optional and disabled by default. If an AECdump is provided and the mic gain simulation is enabled, an extra "level undo" step is performed to virtually restore the unmodified mic signal.
This CL has been ported from https://codereview.webrtc.org/2834643002/.
Bug: webrtc:7494
Change-Id: I0df52b5d45a6bfa1efced980d8d6de5c5d9bed48
Reviewed-on: https://webrtc-review.googlesource.com/2685
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19992}
We are doing some unconventional stuff in jni_generator_helper.h in
order to integrate the Chromium script with WebRTC. Long term, we will
improve this and remove the lint suppressions.
Bug: webrtc:8278
Change-Id: I5d6f0017c4deab4586844647f7cd657641fecbab
Reviewed-on: https://webrtc-review.googlesource.com/3780
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19989}
A value for this flag was derived in RtpReceiverImpl::IncomingRtpPacket.
For audio, it was never used, and for video, it was overridden by
the result from RtpDepacketizer::ParsedPayload.
Bug: webrtc:7135
Change-Id: I597a57ca77d13b9a9145a9ee5e6624c1986777b9
Reviewed-on: https://webrtc-review.googlesource.com/3660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19985}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't
know if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e
Reviewed-on: https://webrtc-review.googlesource.com/2560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19977}
