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}
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}
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}
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}
--- 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}
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}
There was quite a lot of excessive qualification in the test code.
Cleaned up by adding a few using directives.
BUG=None
Change-Id: I03487f457f15f663acf276ce2a4a9cbbc9804101
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170467
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30806}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.
Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame
and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.
Bug: chromium:1013590
Change-Id: I136132c210e5811547f2522ddc371d0acac90664
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161093
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30001}
This makes the WebRTC-KeepAbsSendTimeExtension field trial
always enabled. This means that we no longer avoid sending the
abs-send-time extension if we have negotiated sending of transport
wide sequence numbers.
The field trial WebRTC-FilterAbsSendTimeExtension is introduced to allow
reverting to the previous behavior.
Bug: webrtc:10234
Change-Id: Ifd9761d84dd1fe79af840f98ad0882a2e5adf0b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29999}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.
Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.
Reason for revert: crbug.com/1028937
Original change's description:
> Add support for RtpEncodingParameters::max_framerate
>
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}
TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.
Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
The experiment was extended to support per-codec minimum bitrates
for the following codecs:
* VP8
* VP9
* H.264
The old semantic meaning for the field trial is retained, in that
specifying "br:" applies a minimum bitrate to all codecs. If "br:"
is not specified, the per-codec minimum config is consulted.
Bug: webrtc:11024
Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29450}
This enables creation and removal of receive streams with SSRC 0.
Several related methods, for example SetOutputVolume, still use 0 as a
special value.
Bug: webrtc:8694
Change-Id: I341e6bd6c981c9838997510d8d712ad2948f6460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29398}
Rasmus Brandt