It is now used only by FileRotatingStream.
Bug: webrtc:6424
Change-Id: I216b20baadae836d24c39899efe4cb45c2935f41
Reviewed-on: https://webrtc-review.googlesource.com/4720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20040}
We currently suppress warnings for bad constructs in media/. Still, the
warnings are causing problems when trying to include header files from
this directory. This CL cleans up some of the bad constructs.
Bug: None
Change-Id: I808ad39eb23870d20fa5bb05429b50c9078543ae
Reviewed-on: https://webrtc-review.googlesource.com/4541
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20039}
Previously VideoFrame.Buffers would be converted to I420 if
apply_rotation() is true. With this change the operation is skipped if
the rotation is 0.
Bug: webrtc:7749
Change-Id: I24a1a8801e41d8f415b33fe57fec953b74df7459
Reviewed-on: https://webrtc-review.googlesource.com/4665
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20038}
All previous initialize methods are deprecated and a new initialize
that uses a builder pattern is added. This gives us full control over
the order of initialization.
Bug: webrtc:7474
Change-Id: I006190e50f2e75c5015f0be75b86d367676db2cc
Reviewed-on: https://webrtc-review.googlesource.com/4160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20037}
This reverts commit 57fb3154b5411934b80051ad827db4e54d00f381.
Reason for revert: Breaks jingle_glue in chromium; need to leave candidate.h in place and include the new location until it's fixed.
Original change's description:
> Clean up libjingle API dependencies.
>
> This CL moves candidate.h into the public API, since it has
> been implicitly included before.
>
> This is a straightforward way of solving the circular
> dependencies involving that file. For instance,
> libjingle_peerconnection_api includes candidate.h from
> jsepicecandidate.h, but _api can't depend on rtc_p2p, which
> depends on _api. In fact, _api can't depend on much at all
> since it's a very high level abstraction; instead, things
> should depend on it.
>
> Furthermore, we have the case where deprecated headers
> include headers in internal modules. I just have to turn
> off include checking for those, but that's not a big deal.
>
> This CL punts the problem of callfactoryinterface.h being
> implicitly included, and pulling in most of the call
> module with it. This should be addressed in a follow-up
> CL.
>
> Bug: webrtc:7504
> Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
> Reviewed-on: https://webrtc-review.googlesource.com/2020
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20034}
TBR=phoglund@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org
Change-Id: Ic5c3d0cf0b8c4d48ecbc49efdb76b373e3c950a5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7504
Reviewed-on: https://webrtc-review.googlesource.com/4702
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20036}
This reverts commit 3dc4d4a21f80cdf44c508412d784b254957696eb.
Reason for revert: breaks internal project
Original change's description:
> Move clients of WebRtcSession to use PeerConnection
>
> This change is part of the work to merge WebRtcSession into
> PeerConnection. To make that work easier, this moves all clients
> of WebRtcSession to use shims added to PeerConnection. That way
> when the classes are merged they won't need to be modified.
>
> Bug: webrtc:8183
> Change-Id: I43de7acf7e38c9fcf2dbf55d50eb05e73767c251
> Reviewed-on: https://webrtc-review.googlesource.com/4320
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20030}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: I13f335b24c26753429cd08a4ca3e295eed5660ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8183
Reviewed-on: https://webrtc-review.googlesource.com/4700
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20035}
This CL moves candidate.h into the public API, since it has
been implicitly included before.
This is a straightforward way of solving the circular
dependencies involving that file. For instance,
libjingle_peerconnection_api includes candidate.h from
jsepicecandidate.h, but _api can't depend on rtc_p2p, which
depends on _api. In fact, _api can't depend on much at all
since it's a very high level abstraction; instead, things
should depend on it.
Furthermore, we have the case where deprecated headers
include headers in internal modules. I just have to turn
off include checking for those, but that's not a big deal.
This CL punts the problem of callfactoryinterface.h being
implicitly included, and pulling in most of the call
module with it. This should be addressed in a follow-up
CL.
Bug: webrtc:7504
Change-Id: I1b1729408158418333ccdf702bf529386090f0d7
Reviewed-on: https://webrtc-review.googlesource.com/2020
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20034}
Allow a custom version of audioproc_f in APM-QA.
Bug: webrtc:7494
Change-Id: Id9adffd63927202d868bc2fc8b6a54c8e6b07039
Reviewed-on: https://webrtc-review.googlesource.com/4060
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20033}
See https://chromium-review.googlesource.com/687499 for the corresponding Chromium change.
Bug: None
Change-Id: I23330d161dc60fd4c8681e58ce5a8e20a2b4a3b8
Reviewed-on: https://webrtc-review.googlesource.com/4540
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20032}
This change is part of the work to merge WebRtcSession into
PeerConnection. To make that work easier, this moves all clients
of WebRtcSession to use shims added to PeerConnection. That way
when the classes are merged they won't need to be modified.
Bug: webrtc:8183
Change-Id: I43de7acf7e38c9fcf2dbf55d50eb05e73767c251
Reviewed-on: https://webrtc-review.googlesource.com/4320
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20030}
Function pointer tables require relocations, so this goes into
.data.rel.ro, not .rodata, but this will at least mark the pages
read-only after relocations are resolved.
Bug: None
Change-Id: I8625e7466b2dcadafc4e4e5f9c6eccbd87af7109
Reviewed-on: https://webrtc-review.googlesource.com/4580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20029}
SSL_CIPHER_standard_name is a bit easier to use. BoringSSL has the
strings in the library statically these days. (Turns out that's more
size-efficient than the code to build it up anyway!)
Bug: None
Change-Id: I91ffa725fa716791cdf75d944cf8d9a3e2cb9021
Reviewed-on: https://webrtc-review.googlesource.com/4362
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20028}
Stop using PayloadUnion's public member variables, since a future CL
will make them private.
BUG=webrtc:8159
Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}
I need to replace the audio part of PayloadUnion with SdpAudioFormat,
but that's a non-trivially-deletable class and those just don't work
well in unions, especially unions that don't have a discriminator that
says which member is currently active.
This CL converts the union to a class, adds a discriminator, and
provides accessor functions. CL #2 in the series will change all
outsiders to use the accessors instead of the public member variables
directly, and CL #3 will remove the public member variables. (It needs
to be done in separate steps like this because PayloadUnion is
unfortunately part of the API, and just changing it all in one go
would break users.)
BUG=webrtc:8159
Change-Id: I38c44bbb21a2d38600cff59bf37d8d47dfdbce21
Reviewed-on: https://webrtc-review.googlesource.com/4340
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20025}
This reverts commit 0cbaf1a6f6ad13a25993f6ea3be931894a196834.
Reason for revert: Makes a test flaky:
https://build.chromium.org/p/client.webrtc/builders/Android32%20%28M%20Nexus5X%29/builds/4603
Original change's description:
> Use injectable hardware video decoder/encoder in AppRTCMobile.
>
> 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}
TBR=magjed@webrtc.org,sakal@webrtc.org
Change-Id: I6cb9a10eadb0eff2b85d5028d684746dc69bccfb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7760
Reviewed-on: https://webrtc-review.googlesource.com/4480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20024}
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}
This is useful for various reasons.
Bug: None
Change-Id: I8658f8b19829cc8470789c13ff3af6466f200f00
Reviewed-on: https://webrtc-review.googlesource.com/4383
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20018}
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}
