These are required to avoid compilation errors due to signed/unsigned
comparison, on some toolchains (specifically Clang/Fuchsia).
Bug: chromium:839351
Change-Id: I52e726acd4e8d6744e98d7583bc82fcec81060d9
Reviewed-on: https://webrtc-review.googlesource.com/74100
Commit-Queue: Wez <wez@google.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23109}
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.
Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
The %f -> %g changes are there to make the SimpleStringBuilder behave like stringstream/cout and friends. With %f it prints e.g. 2.2 as 2.200000. This change also makes it use scientific notation for numbers > 1000000, like the other streams.
Bug: webrtc:8982
Change-Id: Id174b02c5776460607f48efc196a08673b67eeb0
Reviewed-on: https://webrtc-review.googlesource.com/67080
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23094}
This deletes the resilienceOn flag in VideoCodecVP8 and VideoCodecVP9.
Instead, the implementations of VP8 and VP9 set resilience mode
internally, based on the configuration of temporal and spatial layers.
The nack_enabled argument to VideoCodecInitializer::SetupCodec becomes
unused with this cl. In a followup, it will be deleted, together with
the corresponding argument to VideoStreamEncoder methods.
An applications which really wants to configure resilience differently
can do that by injecting an EncoderFactory with encoders behaving
as desired.
Bug: webrtc:8830
Change-Id: I9990faf07d3e95c0fb4a56fcc9a56c2005b4a6fa
Reviewed-on: https://webrtc-review.googlesource.com/71380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23025}
The goal is to make these injectable, and only VP8 and VP9 specific
targets should depend on them.
Bug: webrtc:7925
Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2
Reviewed-on: https://webrtc-review.googlesource.com/72082
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23021}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
We only support on (formely kResilientStream) and off (formely
kResilienceOff). The third mode, kResilientFrames, was not
implemented.
Bug: None
Change-Id: Ida82f6a33eda9d943ea70bc8ae4e6bddb720b0e8
Reviewed-on: https://webrtc-review.googlesource.com/71481
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22984}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
The JsepTransport2 is renamed to JsepTransport.
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: Ibd880e7b744bd912336a691309950bc18e42cf62
Reviewed-on: https://webrtc-review.googlesource.com/65786
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22867}
Per-packet info is now signaled in SentPacket to provide useful stats
for bandwidth consumption and overhead analysis in the network stack.
Bug: webrtc:9103
Change-Id: I2b8f6491567d0fa54cc559fc5a96d7aac7d9565e
Reviewed-on: https://webrtc-review.googlesource.com/66281
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22834}
Contents of the template class VideoMediaChannelTest copied into
VideoMediaChannelTest, the only user.
Bug: None
Change-Id: Ie43a7c4bc1e85e2df77361f43776d4d902b74cae
Reviewed-on: https://webrtc-review.googlesource.com/67400
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22822}
Reject the local/remote description trying to change the pre-negotiated
BUNDLE tag.
Reject an answer containing a BUNDLE group that's not a subset of the offered group.
Reject an offer/answer with a BUNDLE group containing a MID that no m= section has.
Reject an answer removes an m= section from an established BUNDLE group without
rejecting it.
Bug: chromium:827917
Change-Id: If334eefb00b1c1c1e24f9afba0cb00b5867f5590
Reviewed-on: https://webrtc-review.googlesource.com/67190
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22813}
This argument was previously used to implement track muting
(black frames) in the video engine, but that now happens in
the VideoTrack/VideoBroadcaster upstream.
Bug: webrtc:6983
Change-Id: Ib721b297d9fbe55b641c56690dbbd37a52edbb2f
Reviewed-on: https://webrtc-review.googlesource.com/67341
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22785}
This allows clients to enable Receiver reference time reports via
PeerConnection.
RRTR is not enabled by default but can be added to SDP string.
Bug: webrtc:9108
Change-Id: I851f0d65152875bf115553a851b839f83e3d241e
Reviewed-on: https://webrtc-review.googlesource.com/66861
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22767}
With the latest usrsctp roll, the MTU value you provide is the space
avaiable for chunks in the packet. We previously specified this to be the
MTU for the entire SCTP packet, so we were logging errors when the SCTP
packets were 12 bytes larger than expected (the size of the SCTP header).
This fix updates our MTU specified to account for the SCTP header size
as well.
Bug: webrtc:9082
Change-Id: Id3bfa839d4e7662230111ebbdf33bd81ccdc7cf4
Reviewed-on: https://webrtc-review.googlesource.com/66943
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22754}
Currently in the SDP we require an a=ssrc line in the m= section in
order for a StreamParams object to be created with that
MediaContentDescription. This change creates a StreamParams object
without ssrcs in the case that a=msid lines are signaled, but ssrcs
are not. When the remote description is set, this allows us to store
the "unsignaled" StreamParams object in the media channel to later
be used when the first packet is received and we create the
receive stream.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib6734abeee62b8ed688a8208722c402134c074ef
Reviewed-on: https://webrtc-review.googlesource.com/63141
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22712}
In previous implementation, the SctpTransport always assumes the
DtlsTransport underneath is non-null, which is not true after switching
to new JsepTransportController model.
This CL adds nullptr when connecting/disconnecting the SctpTransport with
the DtlsTransport.
The "channel" related methods and variables are also renamed.
Bug: chromium:827917, chromium:828220
Change-Id: I95aa2900d23b0885f45500e2c53def771abdccad
Reviewed-on: https://webrtc-review.googlesource.com/66160
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22700}
This is a reland of f351c3408a0c7f695447a2a9f4e6a1719a0d6a26
Reland history:
The original CL broke tests in chromium which were manually tested in
the first reland. Another small fix was added to the reland to fix a
downstream bug, which caused separate tests to fail in chromium.
These were not caught because the chromium trybot was down. These
are temporarily disabled in chrome to allow this change to roll in.
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=deadbeef@webrtc.org
Bug: webrtc:7932, webrtc:7933
Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17
Reviewed-on: https://webrtc-review.googlesource.com/66280
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22699}
This allows us to update downstreams to no longer use the constructor
that includes groupid. This is part of a small cleanup of StreamParams.
Bug: webrtc:9042
Change-Id: I0343fd9213157614023e5990f6e776b4f56de144
Reviewed-on: https://webrtc-review.googlesource.com/63421
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22695}
This reverts commit f351c3408a0c7f695447a2a9f4e6a1719a0d6a26.
Reason for revert: Breaks chromium import
https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/58012
Failin tests:
WebRtcRtpBrowserTest.TrackAddedToSecondStream
WebRtcRtpBrowserTest.TrackSwitchingStream
Original change's description:
> Reland "Adds support for multiple or no media stream ids."
>
> This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
>
> Original change's description:
> > Adds support for multiple or no media stream ids.
> >
> > With Unified Plan SDP semantics, this adds support for specifying
> > either no media stream ids or multiple media stream ids for a
> > transceiver/sender/receiver. This includes serializing/deserializing
> > SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> > <appdata>" line to indicate the no stream case. Note that this does
> > not synchronize between multiple streams, this is still just supported
> > based upon the first media stream id.
> >
> > Bug: webrtc:7932, webrtc:7933
> > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> > Reviewed-on: https://webrtc-review.googlesource.com/61341
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22611}
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
> Reviewed-on: https://webrtc-review.googlesource.com/65560
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22687}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org
Change-Id: I1835419f963762bc308a91d81c423d8e7bf65026
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7932, webrtc:7933
Reviewed-on: https://webrtc-review.googlesource.com/65700
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Commit-Queue: Tomas Gunnarsson <tommi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22690}
This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb
Original change's description:
> Adds support for multiple or no media stream ids.
>
> With Unified Plan SDP semantics, this adds support for specifying
> either no media stream ids or multiple media stream ids for a
> transceiver/sender/receiver. This includes serializing/deserializing
> SDPs with multiple a=msid lines in a m section, or an "a=msid:-
> <appdata>" line to indicate the no stream case. Note that this does
> not synchronize between multiple streams, this is still just supported
> based upon the first media stream id.
>
> Bug: webrtc:7932, webrtc:7933
> Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
> Reviewed-on: https://webrtc-review.googlesource.com/61341
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22611}
Bug: webrtc:7932, webrtc:7933
Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed
Reviewed-on: https://webrtc-review.googlesource.com/65560
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22687}
This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
In preparation for also moving the responsibility for encoder creation.
Bug: webrtc:8830
Change-Id: Ic3d2039a86cd31c1b4157f5df4e97b607c81f1d7
Reviewed-on: https://webrtc-review.googlesource.com/55264
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22630}
Number of temporal layers in screen sharing was unintentionally changed
from 3 to 2 in a796a7ee85b8805a92e21f888a893bef1581bfee.
This changes the value to 3.
Bug: webrtc:9013
Change-Id: I68291b49276afd2689f20d1c3581e149aa6fd610
Reviewed-on: https://webrtc-review.googlesource.com/61860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22621}
This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Revert "Replace BundleFilter with RtpDemuxer in RtpTransport."
>
> This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e.
>
> Reason for revert: Broke chromium tests.
> Original change's description:
> > Replace BundleFilter with RtpDemuxer in RtpTransport.
> >
> > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
> > type-based demuxing. RtpTransport will support MID-based demuxing later.
> >
> > Each BaseChannel has its own RTP demuxing criteria and when connecting
> > to the RtpTransport, BaseChannel will register itself as a demuxer sink.
> >
> > The inheritance model is changed. New inheritance chain:
> > DtlsSrtpTransport->SrtpTransport->RtpTranpsort
> >
> > NOTE:
> > When RTCP packets are received, Call::DeliverRtcp will be called for
> > multiple times (webrtc:9035) which is an existing issue. With this CL,
> > it will become more of a problem and should be fixed.
> >
> > Bug: webrtc:8587
> > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
> > Reviewed-on: https://webrtc-review.googlesource.com/61360
> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22613}
>
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
>
> Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8587
> Reviewed-on: https://webrtc-review.googlesource.com/64860
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22614}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8587
Reviewed-on: https://webrtc-review.googlesource.com/64862
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22615}
This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e.
Reason for revert: Broke chromium tests.
Original change's description:
> Replace BundleFilter with RtpDemuxer in RtpTransport.
>
> BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
> type-based demuxing. RtpTransport will support MID-based demuxing later.
>
> Each BaseChannel has its own RTP demuxing criteria and when connecting
> to the RtpTransport, BaseChannel will register itself as a demuxer sink.
>
> The inheritance model is changed. New inheritance chain:
> DtlsSrtpTransport->SrtpTransport->RtpTranpsort
>
> NOTE:
> When RTCP packets are received, Call::DeliverRtcp will be called for
> multiple times (webrtc:9035) which is an existing issue. With this CL,
> it will become more of a problem and should be fixed.
>
> Bug: webrtc:8587
> Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
> Reviewed-on: https://webrtc-review.googlesource.com/61360
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22613}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8587
Reviewed-on: https://webrtc-review.googlesource.com/64860
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22614}
BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
type-based demuxing. RtpTransport will support MID-based demuxing later.
Each BaseChannel has its own RTP demuxing criteria and when connecting
to the RtpTransport, BaseChannel will register itself as a demuxer sink.
The inheritance model is changed. New inheritance chain:
DtlsSrtpTransport->SrtpTransport->RtpTranpsort
NOTE:
When RTCP packets are received, Call::DeliverRtcp will be called for
multiple times (webrtc:9035) which is an existing issue. With this CL,
it will become more of a problem and should be fixed.
Bug: webrtc:8587
Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
Reviewed-on: https://webrtc-review.googlesource.com/61360
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22613}
With Unified Plan SDP semantics, this adds support for specifying
either no media stream ids or multiple media stream ids for a
transceiver/sender/receiver. This includes serializing/deserializing
SDPs with multiple a=msid lines in a m section, or an "a=msid:-
<appdata>" line to indicate the no stream case. Note that this does
not synchronize between multiple streams, this is still just supported
based upon the first media stream id.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275
Reviewed-on: https://webrtc-review.googlesource.com/61341
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22611}
This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
