1e257cacbf
Migrate media/ to webrtc::Mutex.
...
Bug: webrtc:11567
Change-Id: I69e4a1b37737ac8dd852a032612623c4c4f3a30b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176744
Commit-Queue: Markus Handell <handellm@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31648}
2020-07-07 13:46:47 +00:00
3cb0985983
Inclusive language in //media/engine.
...
Bug: webrtc:11680
Change-Id: I4f21ecaf1e0cc35591ed00d776eb382b868fc076
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178391
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31589}
2020-06-30 13:13:55 +00:00
2a70703eb8
Delete MediaTransportInterface and DatagramTransportInterface
...
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Taylor <deadbeef@webrtc.org >
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31536}
2020-06-17 08:41:14 +00:00
9465978a3b
Remove framemarking RTP extension.
...
BUG=webrtc:11637
Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +00:00
e917379c5b
[Stats] Don't attempt to aggregate empty VideoSenderInfos.
...
This fixes a crash that could happen if substreams exist but there is
no kMedia substream yet. There was an assumption that we either had no
substreams or at least one kMedia substream, but this was not true.
The correct thing to do is to ignore substream stats that are not
associated with any kMedia substream, because we only produce
"outbound-rtp" stats objects for existing kMedia substreams.
A test is added to make sure no stats are returned. Prior to the fix,
this test would crash.
Bug: chromium:1090712
Change-Id: Ib1f8494a162542ae56bdd2df7618775a3473419b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176446
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31442}
2020-06-04 09:03:52 +00:00
a0ff50c031
Reland "Improve outbound-rtp statistics for simulcast"
...
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org >
> > Reviewed-by: Erik Språng <sprang@webrtc.org >
> > Reviewed-by: Henrik Boström <hbos@webrtc.org >
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> > Commit-Queue: Eldar Rello <elrello@microsoft.com >
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Commit-Queue: Henrik Boström <hbos@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31169}
2020-05-05 20:22:19 +00:00
9a925c9ce3
Revert "Improve outbound-rtp statistics for simulcast"
...
This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
Reason for revert: Breaks googRtt in legacy getStats API
Original change's description:
> Improve outbound-rtp statistics for simulcast
>
> Bug: webrtc:9547
> Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> Reviewed-by: Sebastian Jansson <srte@webrtc.org >
> Reviewed-by: Erik Språng <sprang@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Eldar Rello <elrello@microsoft.com >
> Cr-Commit-Position: refs/heads/master@{#31097}
TBR=hbos@webrtc.org ,sprang@webrtc.org ,stefan@webrtc.org ,srte@webrtc.org ,hta@webrtc.org ,elrello@microsoft.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9547
Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31165}
2020-05-05 13:38:51 +00:00
d9255b1840
[getStats] Fix DCHECK crash in MergeInfoAboutOutboundRtpSubstreams().
...
It seems possible that getStats() and merging RTX/FlexFEC substream
stats into media substream stats can race with the creation or
destruction of the media substream that the RTX/FlexFEC substream is
associated with.
In other words, the DCHECK that ensures that there exists a stats object
to merge into is not always valid. Because there is no media stats
object to merge in to, and outbound-rtp stats objects only exists per
media SSRCs, the sensible thing to do is to RTC_LOG and ignore the
substream stats.
Bug: webrtc:11545
Change-Id: I4061d7190da7ab8bd33fa1fd92c9d819f35d76c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31156}
2020-05-04 15:25:34 +00:00
da6cda839d
Improve outbound-rtp statistics for simulcast
...
Bug: webrtc:9547
Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Eldar Rello <elrello@microsoft.com >
Cr-Commit-Position: refs/heads/master@{#31097}
2020-04-17 11:28:00 +00:00
4a5bce96e8
Change to more idiomatic map erase.
...
Bug: webrtc:11477
Tested: JS application with early video.
Change-Id: I2733127744f6c1c32da1acb3533428e451cd65dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173589
Reviewed-by: Tommi <tommi@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31076}
2020-04-15 11:15:17 +00:00
adc4da30f4
[InsertableStreams] Fix video receiver simulcast.
...
Save the frame transformer set on unsignaled receivers, and set the
transformer when the ssrc becomes known.
Pass the receiver's ssrc on registering the transformed frame callback,
to associate separate frame transformer sinks for each receiver.
Bug: chromium:1065838
Bug: chromium:1065838
Change-Id: I2a214bdb6cb9a8012928a03f046f311c344370f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173201
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31051}
2020-04-11 12:04:24 +00:00
16d0d371d5
Apply performance-for-range-copy fixes.
...
This CL has been generated running https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html .
Bug: None
Change-Id: Ia9f6c91776fc8b3ab28fba87ba8ce112f87d5cf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172805
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30996}
2020-04-03 11:36:52 +00:00
de6fa1ef29
Reland "Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams."
...
This is a reland of d335426a39d34389a00f8f7ae652d535f0fa2073.
The revert was premature: the failing tests were known to be flaky
(crbug.com/1066515, crbug.com/1066453, crbug.com/1066407, crbug.com/1066399)
Original change's description:
> Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams.
>
> This CL changes WebRtcVideoChannel::ResetUnsignaledRecvStream so
> that it deletes all default streams created by
> WebRtcVideoChannel::AddRecvStream. This is needed for the case that
> there are lingering default streams, whose SSRCs are different
> from the SSRCs that were subsequently signaled. This can happen
> when there are multiple "m= sections" and the early media is
> sent to an "m= section" that is later not supposed to be the
> sink for that particular SSRC.
>
> Default streams whose SSRC match the subsequently signaled
> SSRC is already handled here: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1386;drc=22387b44ff173d263b434889d394cea90368ab06?originalUrl=https:%2F%2Fcs.chromium.org%2F
>
> Bug: webrtc:11477
> Change-Id: I96ed7e35b4904fb0757fe5824f8afa6f1b9a565e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172622
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30971}
TBR=mflodman@webrtc.org ,hta@webrtc.org
Bug: webrtc:11477
Change-Id: I70b8fa47b4d1d0aa36fed4d8612e13fa7f992925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172782
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30986}
2020-04-02 16:08:26 +00:00
c6b2f34f35
Revert "Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams."
...
This reverts commit d335426a39d34389a00f8f7ae652d535f0fa2073.
Reason for revert: Breaking RTCPeerConnectionTest.GetTrackRemoveStreamAndGCAll.
Original change's description:
> Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams.
>
> This CL changes WebRtcVideoChannel::ResetUnsignaledRecvStream so
> that it deletes all default streams created by
> WebRtcVideoChannel::AddRecvStream. This is needed for the case that
> there are lingering default streams, whose SSRCs are different
> from the SSRCs that were subsequently signaled. This can happen
> when there are multiple "m= sections" and the early media is
> sent to an "m= section" that is later not supposed to be the
> sink for that particular SSRC.
>
> Default streams whose SSRC match the subsequently signaled
> SSRC is already handled here: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1386;drc=22387b44ff173d263b434889d394cea90368ab06?originalUrl=https:%2F%2Fcs.chromium.org%2F
>
> Bug: webrtc:11477
> Change-Id: I96ed7e35b4904fb0757fe5824f8afa6f1b9a565e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172622
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30971}
TBR=brandtr@webrtc.org ,mflodman@webrtc.org ,hta@webrtc.org
Change-Id: I41dc2ea2fc43bb3f7cca2fc5e946c58baa54e00a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11477
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172760
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30979}
2020-04-02 12:26:19 +00:00
d335426a39
Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams.
...
This CL changes WebRtcVideoChannel::ResetUnsignaledRecvStream so
that it deletes all default streams created by
WebRtcVideoChannel::AddRecvStream. This is needed for the case that
there are lingering default streams, whose SSRCs are different
from the SSRCs that were subsequently signaled. This can happen
when there are multiple "m= sections" and the early media is
sent to an "m= section" that is later not supposed to be the
sink for that particular SSRC.
Default streams whose SSRC match the subsequently signaled
SSRC is already handled here: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1386;drc=22387b44ff173d263b434889d394cea90368ab06?originalUrl=https:%2F%2Fcs.chromium.org%2F
Bug: webrtc:11477
Change-Id: I96ed7e35b4904fb0757fe5824f8afa6f1b9a565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172622
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30971}
2020-04-02 09:05:43 +00:00
e1aa22f892
[InsertableStreams] Set video frame transformer if RTP stream already started.
...
Test in https://chromium-review.googlesource.com/c/chromium/src/+/2127927
Bug: chromium:1065836
Change-Id: Idf3f41285e23ac829f69f1bc95b1def3a73af8d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172400
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Marina Ciocea <marinaciocea@webrtc.org >
Commit-Queue: Guido Urdaneta <guidou@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30948}
2020-03-31 14:07:29 +00:00
3e98368ec5
Reland "Distinguish between send and receive codecs"
...
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30932}
2020-03-29 21:03:27 +00:00
f45ca3787f
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
...
--- Background ---
The webrtc::VideoSendStream::StreamStats are converted into
VideoSenderInfo objects which turn into "outbound-rtp" stats objects in
getStats() (or "ssrc" objects in legacy getStats()).
StreamStats are created for each type of substream: RTP media streams,
RTX streams and FlexFEC streams - each with individual packet counters.
The RTX stream is responsible for retransmissions of a referenced media
stream and the FlexFEC stream is responsible for FEC of a referenced
media stream. RTX/FEC streams do not show up as separate objects in
getStats(). Only the media streams become "outbound-rtp" objects, but
their packet and byte counters have to include the RTX and FEC counters.
--- Overview of this CL ---
This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes
StreamStats of all kinds as input, and outputs media-only StreamStats
- incorporating the RTX and FEC counters into the relevant media
StreamStats.
The merged StreamStats objects is a smaller set of objects than the
non-merged counterparts, but when aggregating all packet counters
together we end up with exact same packet and count as before.
Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates
the StreamStats into a single VideoSenderInfo (single "outbound-rtp"),
this CL should not have any observable side-effects. Prior to this CL:
aggregate StreamStats. After this CL: merge StreamStats and then
aggregate them.
However, when simulcast stats are implemented (WIP CL:
https://webrtc-review.googlesource.com/c/src/+/168120 ) each RTP media
stream should turn into an individual "outbound-rtp" object. We will
then no longer aggregate all StreamStats into a single "info". This CL
unblocks simulcast stats by providing StreamStats objects that could be
turned into individual VideoSenderInfos.
--- The Changes ---
1. Methods added to RtpConfig to be able to easily tell the relationship
between RTP, RTX and FEC ssrcs.
2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that
replaces the booleans (is_rtx, is_flexfec).
3. "referenced_media_ssrc" is added to StreamStats, making it possible
to tell which kRtx/kFlexFec stream stats need to be merged with which
kMedia StreamStats.
4. MergeInfoAboutOutboundRtpSubstreams() added and used.
Bug: webrtc:11439
Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30869}
2020-03-24 13:31:54 +00:00
fc29b0ad46
[Stats] Include RTX retransmissions in the VideoSenderInfo.
...
Ignoring retransmissions carried over the RTX stream was a bug. This CL
fixes the bug, so that all retransmissions are accounted for. It also
adds test coverage for this.
This resolves https://crbug.com/webrtc/11440 but does not resolve
https://crbug.com/webrtc/11439 .
Bug: webrtc:11440, webrtc:11439
Change-Id: Ifb10aa60a0f453738aaa30de90eaa5b31f9ec265
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170639
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30822}
2020-03-18 17:25:16 +00:00
0357b3e7b6
RtpTransceiverInterface: add header_extensions_to_offer()
...
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104 .
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30800}
2020-03-16 13:16:42 +00:00
b05ca4b616
Implement new specification for degradation preference
...
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30691}
2020-03-05 14:24:25 +00:00
3f1aee3cbb
Change network_priority from a double to an enum.
...
It can only be one of four possible values, so it never made sense
for it to be a double. Other than the fact that its neighbor
bitrate_priority is a double, and they're both defined as the same enum
in the web spec. However, while bitrate_priority being a double
offers more flexibility than the web spec, network_priority being a
double is only confusing.
Bug: webrtc:5658
Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840
Reviewed-by: Anders Carlsson <andersc@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Taylor <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30685}
2020-03-05 05:42:15 +00:00
24dbb21383
Enable quality scaler for simulcast and SVC if only one stream is active
...
Also, make sure active flags are not lost in simulcast encoder adapter
which is needed in case of simulcast encoder adapter is used.
VP9 libvpx encoder currently ignores scaling setting for SVC, but libvpx
fix is incoming.
TESTED=On a manually patched chrome with singlecast-simulcast vp8 stream.
Bug: webrtc:11396
Change-Id: Ic81f014bec1bdaaf6d5d173743933e5d77d71ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169547
Reviewed-by: Evan Shrubsole <eshr@google.com >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30681}
2020-03-04 15:22:00 +00:00
412a31bbf8
Insert frame transformer between Depacketizer and Decoder.
...
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.
The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130 .
This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Magnus Flodman <mflodman@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00
e77912ba8c
Insert frame transformer between Encoded and Packetizer.
...
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.
The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128 .
Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md
Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk
Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-28 07:43:13 +00:00
cad3e0e2fa
Replace DataSize and DataRate factories with newer versions
...
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format
Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18 16:09:50 +00:00
e45763139c
Save custom parameters provided by the SdpVideoFormat when requesting an encoder switch.
...
Bug: webrtc:11341
Change-Id: I1079c4ec021eb3939df4c92d2e1a48874e854dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168645
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30541}
2020-02-18 11:09:49 +00:00
06901cfb04
Use absl::c_any_of instead of a manual for-loop to finding an active encoding
...
Bug: webrtc:11319
Change-Id: I00eff8dd1d595570b9b2798a27514ec16fde4bf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168646
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Oskar Segersvärd <oseg@google.com >
Cr-Commit-Position: refs/heads/master@{#30538}
2020-02-18 10:34:18 +00:00
dc81e11f96
Use webrtc::DataRate when referring to bitrates
...
Change-Id: I1ff344fa1ae302e036f91db19d073e4c9829825f
Bug: webrtc:9709
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168498
Commit-Queue: Oskar Segersvärd <oseg@google.com >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30517}
2020-02-13 13:26:49 +00:00
e2b466e925
Stop advertising generic frame descriptor v1
...
it is deprecated in favor of dependency descriptor rtp header extension
which is a later version of the generic frame descriptor.
Bug: webrtc:11358
Change-Id: I95062885dd204c9afc096a3284df8f66b05998b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168497
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30508}
2020-02-12 16:16:58 +00:00
03d909634b
Ensure that the first active layer isn't disabled by too low input resolution
...
If e.g. CPU adaptation reduces input video size too much, video pipeline would
reduce the number of used simulcast streams/spatial layers. This may result in
disabled video if some streams are disabled by Rtp encoding parameters API.
Bug: webrtc:11319
Change-Id: Id7f157255599dcb6f494129b83477cda4bea982a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168480
Reviewed-by: Evan Shrubsole <eshr@google.com >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30498}
2020-02-11 14:57:51 +00:00
9b05803e19
Implement injectable EncoderSelectorInterface and wire it up in the VideoStreamEncoder.
...
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.
Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30490}
2020-02-10 12:12:47 +00:00
8e8b36a94a
Revert "Reland "Reland "Reland "Distinguish between send and receive codecs""""
...
This reverts commit 184ea66aed43161f05d80fbb74183a2efccca352.
Reason for revert: Breaks downstream projects.
TBR=steveanton@webrtc.org
Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.
>
> Reason for revert: Keep logic as is.
>
> Original change's description:
> > Revert "Reland "Reland "Distinguish between send and receive codecs"""
> >
> > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
> >
> > Reason for revert: Breaks perf test on iOS.
> >
> > Original change's description:
> > > Reland "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> > >
> > > Reason for revert: Flaky test in Chromium fixed.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive codecs""
> > > >
> > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > > >
> > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > > >
> > > > > Reason for revert: Fixed negotiation of send-only clients.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive codecs"
> > > > > >
> > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > > >
> > > > > > Reason for revert: breaks negotiation with send-only clients
> > > > > >
> > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive codecs
> > > > > > >
> > > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > > to be able to keep track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org >
> > > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > > >
> > > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > > >
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > > >
> > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > >
> > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30360}
> > >
> > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30367}
> >
> > TBR=steveanton@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30373}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30415}
TBR=steveanton@webrtc.org ,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ice25339e7dfb9fc75049bd207d097b0910bd4446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168341
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30484}
2020-02-07 15:11:08 +00:00
72859e5e15
Make RtpEncodingParameters to not reverse active flags order
...
Bug: webrtc:11319
Change-Id: If63db02d282ee622c12405f85c0fbae1ba13fcb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168196
Reviewed-by: Florent Castelli <orphis@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30459}
2020-02-05 17:36:26 +00:00
184ea66aed
Reland "Reland "Reland "Distinguish between send and receive codecs"""
...
This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.
Reason for revert: Keep logic as is.
Original change's description:
> Revert "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
>
> Reason for revert: Breaks perf test on iOS.
>
> Original change's description:
> > Reland "Reland "Distinguish between send and receive codecs""
> >
> > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> >
> > Reason for revert: Flaky test in Chromium fixed.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > >
> > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > >
> > > > Reason for revert: Fixed negotiation of send-only clients.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > >
> > > > > Reason for revert: breaks negotiation with send-only clients
> > > > >
> > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive codecs
> > > > > >
> > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > to be able to keep track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > >
> > > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > > >
> > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > >
> > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > >
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > >
> > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30360}
> >
> > TBR=steveanton@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30367}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
> Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30373}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Bug: chromium:1029737
Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30415}
2020-01-29 18:53:54 +00:00
a104ceb0ce
Revert "Reland "Reland "Distinguish between send and receive codecs"""
...
This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
Reason for revert: Breaks perf test on iOS.
Original change's description:
> Reland "Reland "Distinguish between send and receive codecs""
>
> This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
>
> Reason for revert: Flaky test in Chromium fixed.
>
> Original change's description:
> > Revert "Reland "Distinguish between send and receive codecs""
> >
> > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> >
> > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive codecs"
> > >
> > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > >
> > > Reason for revert: Fixed negotiation of send-only clients.
> > >
> > > Original change's description:
> > > > Revert "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > >
> > > > Reason for revert: breaks negotiation with send-only clients
> > > >
> > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive codecs
> > > > >
> > > > > Even though send and receive codecs may be the same, they might have
> > > > > different support in HW. Distinguish between send and receive codecs
> > > > > to be able to keep track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > >
> > > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > > >
> > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Commit-Queue: Steve Anton <steveanton@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > >
> > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > >
> > >
> > > Bug: chromium:1029737
> > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30348}
> >
> > TBR=steveanton@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30360}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
> Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30367}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30373}
2020-01-24 16:44:17 +00:00
9bac68c0cc
Reland "Reland "Distinguish between send and receive codecs""
...
This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
Reason for revert: Flaky test in Chromium fixed.
Original change's description:
> Revert "Reland "Distinguish between send and receive codecs""
>
> This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
>
> Reason for revert: Breaks Chromium import due to flaky test in Chromium.
>
> Original change's description:
> > Reland "Distinguish between send and receive codecs"
> >
> > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> >
> > Reason for revert: Fixed negotiation of send-only clients.
> >
> > Original change's description:
> > > Revert "Distinguish between send and receive codecs"
> > >
> > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > >
> > > Reason for revert: breaks negotiation with send-only clients
> > >
> > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > >
> > > Original change's description:
> > > > Distinguish between send and receive codecs
> > > >
> > > > Even though send and receive codecs may be the same, they might have
> > > > different support in HW. Distinguish between send and receive codecs
> > > > to be able to keep track of which codecs have HW support.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > >
> > > TBR=steveanton@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Commit-Queue: Steve Anton <steveanton@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30292}
> >
> > TBR=steveanton@webrtc.org ,kron@webrtc.org
> >
> >
> > Bug: chromium:1029737
> > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30348}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
> Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30360}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30367}
2020-01-23 23:02:59 +00:00
00a30873c4
Revert "Reland "Distinguish between send and receive codecs""
...
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
Reason for revert: Breaks Chromium import due to flaky test in Chromium.
Original change's description:
> Reland "Distinguish between send and receive codecs"
>
> This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Commit-Queue: Steve Anton <steveanton@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30292}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30348}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30360}
2020-01-23 13:10:53 +00:00
133bf2bd28
Reland "Distinguish between send and receive codecs"
...
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org ,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Steve Anton <steveanton@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30292}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30348}
2020-01-22 13:55:41 +00:00
1acdc748ac
Split up EncoderStreamFactory::CreateEncoderStreams in two.
...
Motivation: https://google.github.io/styleguide/cppguide.html#Write_Short_Functions
This is a pure clean up CL, that should have no functional implications.
Bug: webrtc:11297
Change-Id: I077a8b52254a936b61d1fda94e8cfc39e8cf1294
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166883
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30337}
2020-01-21 14:34:39 +00:00
e57b266a20
Revert "Distinguish between send and receive codecs"
...
This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
Reason for revert: breaks negotiation with send-only clients
(webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
(peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
(peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30284}
TBR=steveanton@webrtc.org ,kron@webrtc.org
Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30292}
2020-01-17 02:47:23 +00:00
c0f25cf762
Distinguish between send and receive codecs
...
Even though send and receive codecs may be the same, they might have
different support in HW. Distinguish between send and receive codecs
to be able to keep track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
Commit-Queue: Johannes Kron <kron@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30284}
2020-01-16 15:42:05 +00:00
b2b2031457
Concatenate string literals at compile time.
...
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
f5ecb5f22e
Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""
...
This reverts commit 9cad4dccc96f09d76dce3766a076052df7d75ab8.
Reason for revert: Breaks downstream tests.
Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive video codecs"""
>
> This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c
>
> This CL lands all code except the code that activates the change,
> see media/engine/webrtc_video_engine.cc
> Once downstream projects are fixed, there will be a one-line change to
> activate the change to distinguish between send and receive video codecs.
>
> Original change's description:
> > Reland "Reland "Distinguish between send and receive video codecs""
> >
> > This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> > >
> > > Reason for revert: Downstream test updated.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive video codecs""
> > > >
> > > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > > >
> > > > Reason for revert: Breaks another downstream test.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > > >
> > > > > Reason for revert: Downstream tests have been updated.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive video codecs"
> > > > > >
> > > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > > >
> > > > > > Reason for revert: Breaks downstream test.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive video codecs
> > > > > > >
> > > > > > > Even though send and receive codecs are the same,
> > > > > > > they might have different support in HW.
> > > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > > track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > > >
> > > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > > >
> > > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > > >
> > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > >
> > > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30079}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30097}
> >
> > Bug: chromium:1029737
> > Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30120}
>
> Bug: chromium:1029737
> Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30219}
TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: I377f82866e56862f57383f96a3b96719344eef9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30225}
2020-01-13 09:03:37 +00:00
9cad4dccc9
Reland "Reland "Reland "Distinguish between send and receive video codecs"""
...
This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c
This CL lands all code except the code that activates the change,
see media/engine/webrtc_video_engine.cc
Once downstream projects are fixed, there will be a one-line change to
activate the change to distinguish between send and receive video codecs.
Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
>
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
>
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are the same,
> > > > > > they might have different support in HW.
> > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30097}
>
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30120}
Bug: chromium:1029737
Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Anders Carlsson <andersc@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30219}
2020-01-10 23:37:11 +00:00
873610ca68
Fix updating degradation preference in SetRtpParameters.
...
Degradation preference could be changed before video send stream
is configured which would cause a crash.
Bug: None
Change-Id: If970e66fba0b9fdb9da789066861d919874de119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164463
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30141}
2020-01-03 11:16:54 +00:00
b5159fe4a7
Revert "Reland "Reland "Distinguish between send and receive video codecs"""
...
This reverts commit 4e64e605894df287178c5a1b537fbe859b7d420c.
Reason for revert: breaks a bunch of WebRtcBrowserTests on Win: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/4843
Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
>
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
>
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are the same,
> > > > > > they might have different support in HW.
> > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30097}
>
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30120}
TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
Change-Id: I709ee0eb6246aa79dde3aacfc4c47e070c4e90ea
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162904
Reviewed-by: Olga Sharonova <olka@webrtc.org >
Commit-Queue: Olga Sharonova <olka@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30122}
2019-12-20 13:57:12 +00:00
4e64e60589
Reland "Reland "Distinguish between send and receive video codecs""
...
This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
Original change's description:
> Reland "Distinguish between send and receive video codecs"
>
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
>
> Reason for revert: Downstream test updated.
>
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> >
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> >
> > Reason for revert: Breaks another downstream test.
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > >
> > > Reason for revert: Downstream tests have been updated.
> > >
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > >
> > > > Reason for revert: Breaks downstream test.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > >
> > > > > Even though send and receive codecs are the same,
> > > > > they might have different support in HW.
> > > > > Distinguish between send and receive codecs to be able to keep
> > > > > track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > >
> > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > >
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> >
> > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30079}
>
> TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30097}
Bug: chromium:1029737
Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Anders Carlsson <andersc@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30120}
2019-12-20 11:44:42 +00:00
f9d92ed2c8
Revert "Reland "Distinguish between send and receive video codecs""
...
This reverts commit 77eb338ae48acb0cb1437da05d86941bb4063228.
Reason for revert: Speculative revert, as it seems to have broken webrtc-importer
Original change's description:
> Reland "Distinguish between send and receive video codecs"
>
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
>
> Reason for revert: Downstream test updated.
>
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> >
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> >
> > Reason for revert: Breaks another downstream test.
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > >
> > > Reason for revert: Downstream tests have been updated.
> > >
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > >
> > > > Reason for revert: Breaks downstream test.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > >
> > > > > Even though send and receive codecs are the same,
> > > > > they might have different support in HW.
> > > > > Distinguish between send and receive codecs to be able to keep
> > > > > track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > >
> > > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > > >
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> >
> > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> >
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30079}
>
> TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> Reviewed-by: Steve Anton <steveanton@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30097}
TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
Change-Id: I73d4fe3bb18e40a01f1b1b0c71f9dc7b85c513b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162208
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30100}
2019-12-16 15:28:41 +00:00
77eb338ae4
Reland "Distinguish between send and receive video codecs"
...
This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
Reason for revert: Downstream test updated.
Original change's description:
> Revert "Reland "Distinguish between send and receive video codecs""
>
> This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
>
> Reason for revert: Breaks another downstream test.
>
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> >
> > Reason for revert: Downstream tests have been updated.
> >
> > Original change's description:
> > > Revert "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > >
> > > Reason for revert: Breaks downstream test.
> > >
> > > Original change's description:
> > > > Distinguish between send and receive video codecs
> > > >
> > > > Even though send and receive codecs are the same,
> > > > they might have different support in HW.
> > > > Distinguish between send and receive codecs to be able to keep
> > > > track of which codecs have HW support.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > >
> > > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> > >
> > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > > Cr-Commit-Position: refs/heads/master@{#30042}
> >
> > TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > Reviewed-by: Johannes Kron <kron@webrtc.org >
> > Reviewed-by: Steve Anton <steveanton@webrtc.org >
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org >
> > Commit-Queue: Johannes Kron <kron@webrtc.org >
> > Cr-Commit-Position: refs/heads/master@{#30078}
>
> TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
>
> Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> Reviewed-by: Johannes Kron <kron@webrtc.org >
> Commit-Queue: Johannes Kron <kron@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#30079}
TBR=steveanton@webrtc.org ,andersc@webrtc.org ,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
Reviewed-by: Johannes Kron <kron@webrtc.org >
Reviewed-by: Anders Carlsson <andersc@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Johannes Kron <kron@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30097}
2019-12-16 14:03:46 +00:00
