This is a reland of bd33ce26202272177af6c52e195e7c13f0d1bf23
Now it doesn't apply flags_compatibility to Android, because the device runner actually requires it to be dashes (so it can intercept the flag and substitute it with an Android-local file path), but that's OK because the runner also already passes the flag with underscores: https://cs.chromium.org/?q=%22--isolated_script_test_perf_output%22
Original change's description:
> Reland "Add wrapper to normalize flags."
>
> This is a reland of 642a49d1eb20b8c5744e745de79ddb585e0f7472
>
> The change has the same effect but is now implemented through mb, rather than specifying a 'script', so that Android's special handling is not skipped.
>
> Original change's description:
> > Add wrapper to normalize flags.
> >
> > Bug: None
> > Change-Id: I9d43602cc66198a29dbc0e7586d948ee76c5ec84
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145204
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28532}
>
> Bug: webrtc:10616
> Change-Id: I60ebd4891dbe8de18c653f8af88181ea966307de
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145409
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28572}
Bug: webrtc:10616
Change-Id: I56aae5475aed62f069c5cecc01b75d7d6ffcf568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145920
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28589}
# NOTRY: All green but win_chromium_compile which fails because
# of unrelated issues
NOTRY=true
Bug: webrtc:10548
Change-Id: I7b6987e7583801d89b91f0e6145b4f1205e30a2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145726
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28579}
This CL fixes a bug affecting the new PacedSender code path (not yet
active): keepalive packets will always be sent, even if sending is
paused or network is congested.
Mainly though, it refactors all the PacedSender unittests and makes
sure all of them (as applicable) are run using both the old and the new
code paths.
Bug: webrtc:10633
Change-Id: I531939b719a11147a2a7dbab77dfcccaf7a42d83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145721
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28578}
Prevent clang-format to touch these two files,
as the result doesn't honor ColumnLimit setting.
Bug: webrtc:9340
Change-Id: I9b692a82df5385fa2d1216d915898439234b34b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#28577}
New PacedSender code path should have the transport sequence numbers
start at 1, just like before. At least some integration tests seem to
depend on this behavior so just to make sure there aren't any parts
doing the same that will break, let's keep the old behavior.
Bug: webrtc:10633
Change-Id: I87290fbfd4ecb2ef04bd5e49a955d0b2cc8d59b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145722
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28573}
This is a reland of 642a49d1eb20b8c5744e745de79ddb585e0f7472
The change has the same effect but is now implemented through mb, rather than specifying a 'script', so that Android's special handling is not skipped.
Original change's description:
> Add wrapper to normalize flags.
>
> Bug: None
> Change-Id: I9d43602cc66198a29dbc0e7586d948ee76c5ec84
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145204
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28532}
Bug: webrtc:10616
Change-Id: I60ebd4891dbe8de18c653f8af88181ea966307de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145409
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28572}
Changing the ssrc for a module is intended to be removed, and will in
the future require creating a new instance.
Bug: webrtc:10774
Change-Id: Ie96daa4a8cf00223ea040509037582f6b1c8eb19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145205
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28571}
On Win8.1 and Win10 when the window is occluded the cropping capturer
falls back to the PrintWindow API. But without a special flag it
fails when trying to capture apps that are using DirectComposition.
So just pass this undocumented flag named PW_RENDERFULLCONTENT to fix
the fallback case of the cropping capturer.
Due to new app framing on Win8 and Win10 the shadow of the window
are captured as black like if for the maximize case on Win7. So
just use the utility function webrtc::GetCroppedWindowRect and
remove the local GetWindowDrawableRect helper. The former returns
the same result as the later on Win7 so no real change is made here,
just that we make the WindowCapturerWin compatible with newer Windows.
Bug: webrtc:10734
Change-Id: Idb793ca0691261042569c30410669c4a5ad0c8ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144960
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28570}
1. Propagate sent notification for all packets, including RTCP
2. Add Field Trial to disable datagram => RTCP feedback loop translation (for tests and experiments only, because having two feedback loops add extra overhead).
Bug: webrtc:9719
Change-Id: Ia3143cc79d127ae331210c86d6675d6e778e962b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145460
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28568}
This reverts commit 11820502b840f34aa9fccb1c273cefbcc457c962.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Make new pacer padding more like old one"
>
> This reverts commit bb7727211c535f8a9dce27891941b52b6ea8e750.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Make new pacer padding more like old one
> >
> > The (currently unused) new pacer code path was implemented with what
> > was intended as a more careful padding strategy.
> > Unfortunately this doesn't work as well as expected due to the fact
> > that the padding budget cannot build up underuse.
> >
> > I made another CL that could fix that, but I think it adds complexity
> > for dubious gains. It also will make it more difficult to find any
> > potential regression when switching to the new path, should one occur.
> > See https://webrtc-review.googlesource.com/c/src/+/144563
> >
> > Therefore, this CL makes the new code path choose RTX payload in the
> > same way as is currently done.
> >
> > Bug: webrtc:10633
> > Change-Id: If2115d4fa7463add959faa77c63101286c27e0f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145202
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28537}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org
>
> Change-Id: I99b72858414e0a245da596d94694449da88fd626
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10633
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145324
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28550}
TBR=mbonadei@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Change-Id: I1fff79d75dc888921c6fbfc7f3980395a67e1c1a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145403
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28565}
This reverts commit 66147e892dd6b7b1beaddbcab456a1ce28b2ad22.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Optimize PacketRouter/RTPSender interactions."
>
> This reverts commit 6f129b3b7605dc69c8c188ca02d133250130570e.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Optimize PacketRouter/RTPSender interactions.
> >
> > The legacy code-path uses a hashmap as cache in order to speed up
> > finding the right rtp module to send on. The new path should use that
> > as well.
> > In addition, there are checks that verify if an RTP module can send
> > padding, in some cases payload based. These result in a number of
> > calls to methods in RTPSender requiring its lock to be taken. This CL
> > introduces a combined SupportsPadding() check method which performs
> > all those checks in one go.
> >
> > Bug: None
> > Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28535}
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I8499dc0fd6e6d0b9fa7a0886c8754655e5589780
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145326
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28552}
TBR=mbonadei@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I3bff3ecb2b776e30f77c1884f6faa72b21788017
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145401
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28563}
This reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f.
Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169.
Original change's description:
> Revert "Reland "Add ability to set RTCP sender ssrc at construction time""
>
> This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
>
> Reason for revert: Speculative revert (some perf test are failing)
>
> Original change's description:
> > Reland "Add ability to set RTCP sender ssrc at construction time"
> >
> > This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
> >
> > Patch set 1 is the original CL.
> > Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> > if either current SSRC is 0 or if the SSRC is identical to the current
> > one. If so, don't schedule an early report.
> > This prevents a regression in which audio jitter became too low(?)
> >
> > Original change's description:
> > > Add ability to set RTCP sender ssrc at construction time
> > >
> > > Bug: webrtc:10774
> > > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#28506}
> >
> > Bug: webrtc:10774
> > Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28520}
>
> TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10774
> Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28555}
TBR=mbonadei@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
Change-Id: I2e5c17e8edfd938424f901222158643baa04866e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10774
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28562}
This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f.
Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
Original change's description:
> Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
>
> This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
>
> Bug: webrtc:10668
> Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28434}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
Bug: webrtc:10668, chromium:982260
Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28561}
This should fix this error:
In file included from ../../api/audio/echo_canceller3_config_json.cc:19:
In file included from ../..\rtc_base/strings/json.h:18:
In file included from ../../third_party/jsoncpp/source/include\json/json.h:9:
In file included from ../../third_party/jsoncpp/source/include\json/autolink.h:9:
../../third_party/jsoncpp/source/include\json/config.h(126,5): error: '__GNUC__' is not defined, evaluates to 0 [-Werror,-Wundef]
Maybe we could also fix it upstream.
Bug: webrtc:10770
Change-Id: I68434c6957189f680625bd2f2b85422965a89785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145322
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28558}
Code is not used in production. Need this change for a local test.
Using TBR.
TBR: henrik.lundin
Bug: webrtc:9265
Change-Id: I9f0cb265a51507de59ef2d7fd151465133687525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145330
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28557}
This partially reverts these 2 CLs:
1) Reland "Copy video frames metadata between encoded and plain frames in one place"
https://webrtc.googlesource.com/src/+/2ebf5239782bf6b46d4aa812f34fa9f9e5a02be9
2) Don't copy video frame metadata in each encoder/decoder
https://webrtc.googlesource.com/src/+/ab62b2ee51e622be6d0aade15e87e927fa60e6f2
The problem with them were that ColorSpace was made to always be copied from the
EncodedImage in the GenericDecoder, which overwrote ColorSpace information from
the decoder.
If decoder applied color space transition or bitstream color space information
was different from the WebRTC signaled one, the incorrect color space data were
passed to the renderer.
This CL removes introduced change regarding color space data: GenericDecoder
doesn't copy or store it and software decoders are restored to copy it.
Relevant tests are also removed.
Bug: chromium:982486
Change-Id: I989e01476ff7f7df376c05578ab8f540b95a1dd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145323
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28556}
This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Reland "Add ability to set RTCP sender ssrc at construction time"
>
> This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
>
> Patch set 1 is the original CL.
> Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> if either current SSRC is 0 or if the SSRC is identical to the current
> one. If so, don't schedule an early report.
> This prevents a regression in which audio jitter became too low(?)
>
> Original change's description:
> > Add ability to set RTCP sender ssrc at construction time
> >
> > Bug: webrtc:10774
> > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28506}
>
> Bug: webrtc:10774
> Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28520}
TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10774
Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28555}
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.
Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
This reverts commit 6f129b3b7605dc69c8c188ca02d133250130570e.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Optimize PacketRouter/RTPSender interactions.
>
> The legacy code-path uses a hashmap as cache in order to speed up
> finding the right rtp module to send on. The new path should use that
> as well.
> In addition, there are checks that verify if an RTP module can send
> padding, in some cases payload based. These result in a number of
> calls to methods in RTPSender requiring its lock to be taken. This CL
> introduces a combined SupportsPadding() check method which performs
> all those checks in one go.
>
> Bug: None
> Change-Id: I2d18d0d6e7d8cfe92c81d08cef248a4daa7dcd4b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144780
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28535}
TBR=asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
Change-Id: I8499dc0fd6e6d0b9fa7a0886c8754655e5589780
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145326
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28552}
This reverts commit bb7727211c535f8a9dce27891941b52b6ea8e750.
Reason for revert: Speculative revert (some perf test are failing)
Original change's description:
> Make new pacer padding more like old one
>
> The (currently unused) new pacer code path was implemented with what
> was intended as a more careful padding strategy.
> Unfortunately this doesn't work as well as expected due to the fact
> that the padding budget cannot build up underuse.
>
> I made another CL that could fix that, but I think it adds complexity
> for dubious gains. It also will make it more difficult to find any
> potential regression when switching to the new path, should one occur.
> See https://webrtc-review.googlesource.com/c/src/+/144563
>
> Therefore, this CL makes the new code path choose RTX payload in the
> same way as is currently done.
>
> Bug: webrtc:10633
> Change-Id: If2115d4fa7463add959faa77c63101286c27e0f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145202
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28537}
TBR=sprang@webrtc.org,stefan@webrtc.org
Change-Id: I99b72858414e0a245da596d94694449da88fd626
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10633
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145324
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28550}
default connection address is a hostname candidate.
Using a FQDN in the c= line has caused an inter-op issue with Firefox
when hostname candidates are the only candidates gathered when forming
the media sections. To address this issue, we use 0.0.0.0:9 when a
hostname candidate would be used to populate the c= and the m= lines.
The SDP grammar related to ICE candidates has been moved out of RFC8445,
and is currently defined in draft-ietf-mmusic-ice-sip-sdp. A FQDN
address must not be used in the connection address attribute per the
latest draft, if the ICE agent generates local candidates. Also, the
wildcard addresses (0.0.0.0 or ::) with port 9 are given the exception
as the connection address that will not result in an ICE mismatch. We
thus adopt the aforementioned solution after combining these
considerations.
Bug: chromium:927309, chromium:982108
Change-Id: I3df2db0f154276da39f99650289cf81baa677e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28547}
This reverts commit 7325bc3917e6dd4c92e7a18fd879ba91f0b2851f.
Reason for revert: FecTest.UlpfecTest is consistently failing.
Original change's description:
> Refactor FEC code to use COW buffers
>
> This refactoring helps to reduce unnecessary memcpy calls on the receive
> side.
>
> This CL is the first stage of refactoring: it only replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| and does
> necessary changes.
>
> A follow-up CL will remove length field of the Packet class.
>
>
> Bug: webrtc:10750
> Change-Id: Ie233da83ff33f6370f511955e4c65d59522389a7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144881
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28539}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org,titovartem@webrtc.org
Change-Id: I07c34256a76174f09a0d27eacbae6488e66f4b43
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145340
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28545}
