These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
Update libwertc AudioRtpSender::SetAudioSend with WEBRTC_WEBKIT_BUILD
This only introduces the WEBRTC_WEBKIT BUILD, inspired by WEBRTC_CHROMIUM_BUILD
macro. It is only defined by Webkit libwebrtc build system.
https://trac.webkit.org/changeset/210977
BUG=webrtc:7039
Review-Url: https://codereview.webrtc.org/2651273003
Cr-Commit-Position: refs/heads/master@{#16432}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Previously in the spec, there was a createDtmfSender method on
PeerConnection, but that's been replaced by a "dtmf" attribute
on RtpSender, which allows getting a DTMF sender without having
an audio track.
This also simplifies the code slightly, since tracks are now not
necessary for identification.
BUG=webrtc:4180
Review-Url: https://codereview.webrtc.org/2666853002
Cr-Commit-Position: refs/heads/master@{#16409}
This should help pave the way for injectable audio codecs, since
external implementations need to be able to signal arbitrary fmtp
parameters.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2661453003
Cr-Commit-Position: refs/heads/master@{#16360}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.
webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.
Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
https://codereview.webrtc.org/2514883002/ changed and moved these targets around but did not add public dependencies for the fallbacks, which causes gn gen --check a lot of anger.
NOTRY=true # Only build changes and windows bots are cranky atm.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2651663002
Cr-Commit-Position: refs/heads/master@{#16214}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2eeTBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}
DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.
BUG=none
Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
Reason for revert:
Failed the memory check.
May need to fix the memory leak.
Original issue's description:
> make the DtlsTransportWrapper inherit form DtlsTransportInternal
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2606123002
> Cr-Commit-Position: refs/heads/master@{#16160}
> Committed: 5aed06c8d3TBR=deadbeef@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2639203004
Cr-Commit-Position: refs/heads/master@{#16162}
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.
In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using
git grep -l ' ASSERT(' | grep -v common.h | \
xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
Previously: Failed to setup RTCP mux filter.
Now: rtcpMuxPolicy is 'require', but media description does not
contain 'a=rtcp-mux'.
BUG=webrtc:6966
Review-Url: https://codereview.webrtc.org/2622553003
Cr-Commit-Position: refs/heads/master@{#16062}
Previously, BaseChannel supported a "no RTCP" mode, which wasn't
being used any more and is being deleted.
Also, "RTCP mux required" previously worked by calling "ActivateRtcpMux"
after construction. Now it works by explicitly passing a
"require_rtcp_mux" parameter into the constructor.
BUG=None
Review-Url: https://codereview.webrtc.org/2622613004
Cr-Commit-Position: refs/heads/master@{#16045}
The BaseChannel can set the transport directly without depending on
TransportController.
When initializing the network of the BaseChannel, the ChannelManager will
create TransportChannels with the TransportController.
When enabling bundling, WebRtcSession will get or create TransportChannels
with the TransportController.
When a TransportChannel of the BaseChannel needs to be destroyed, it will
fire a signal to notify the WebRtcSession.
BUG=none.
Review-Url: https://codereview.webrtc.org/2614263002
Cr-Commit-Position: refs/heads/master@{#16043}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
This cl was produced by
git grep -l 'ASSERT(false)' |\
xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'
followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.
This is a step towards deletion of base/common.h.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Original-Commit-Position: refs/heads/master@{#15906}
Committed: 67b3bbe639
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15973}
Reason for revert:
Hitting DCHECK in chromium's WebrtcTransportTest.TerminateDataChannel and WebrtcTransportTest.DataStreamLate. Will investigate and reland.
Original issue's description:
> Separating SCTP code from BaseChannel/MediaChannel.
>
> The BaseChannel code is geared around RTP; the presence of media engines,
> send and receive streams, SRTP, SDP directional attribute negotiation, etc.
> It doesn't make sense to use it for SCTP as well. This separation should make
> future work both on BaseChannel and the SCTP code paths easier.
>
> SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
> directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
> doesn't get confused with webrtc::DataChannel any more.
>
> Beyond just moving code around, some consequences of this CL:
> - We'll now stop using the worker thread for SCTP. Packets will be
> processed right on the network thread instead.
> - The SDP directional attribute is ignored, as it's supposed to be.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2564333002
> Cr-Commit-Position: refs/heads/master@{#15906}
> Committed: 67b3bbe639TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2614813003
Cr-Commit-Position: refs/heads/master@{#15908}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15906}
Reason for revert:
Doing a reland where systeminfo.cc includes basictypes.h so that CPU_X86 etc. are defined when they are checked/used.
Original issue's description:
> Revert of Replace basictypes.h with stdint.h for int_t types. (patchset #1 id:1 of https://codereview.webrtc.org/2604043002/ )
>
> Reason for revert:
> Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
>
> Original issue's description:
> > Replace basictypes.h with stdint.h for int_t types.
> >
> > Removes basictypes.h for types that only makes use of it for fixed-size-int
> > typedefs and replaces it with stdint.h.
> >
> > BUG=webrtc:6853
> > R=tommi@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2604043002
> > Cr-Commit-Position: refs/heads/master@{#15867}
> > Committed: 7fd1a75300
>
> TBR=tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6853
>
> Review-Url: https://codereview.webrtc.org/2603203003
> Cr-Commit-Position: refs/heads/master@{#15869}
> Committed: 7eb0e23bcf
BUG=webrtc:6853
TBR=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2609783002
Cr-Commit-Position: refs/heads/master@{#15873}
Reason for revert:
Very likely cause of Chromium import bot breakage (unused function '__cpuid'), TBD why.
Original issue's description:
> Replace basictypes.h with stdint.h for int_t types.
>
> Removes basictypes.h for types that only makes use of it for fixed-size-int
> typedefs and replaces it with stdint.h.
>
> BUG=webrtc:6853
> R=tommi@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2604043002
> Cr-Commit-Position: refs/heads/master@{#15867}
> Committed: 7fd1a75300TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6853
Review-Url: https://codereview.webrtc.org/2603203003
Cr-Commit-Position: refs/heads/master@{#15869}
Removes basictypes.h for types that only makes use of it for fixed-size-int
typedefs and replaces it with stdint.h.
BUG=webrtc:6853
R=tommi@webrtc.org
Review-Url: https://codereview.webrtc.org/2604043002
Cr-Commit-Position: refs/heads/master@{#15867}
This allows building without SCTP support (and even building/running
tests). The "HAVE_SCTP" define has been functional for a while, but there
wasn't any easy way to turn it on/off.
NOTRY=True
BUG=webrtc:6933
Review-Url: https://codereview.webrtc.org/2593313002
Cr-Commit-Position: refs/heads/master@{#15763}
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".
BUG=None
Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15453}
Reason for revert:
Deletion of transport.h broke downstream builds.
Going to reland with transport.h containing enums/etc.
Original issue's description:
> Refactoring that removes P2PTransport and DtlsTransport classes.
>
> Their base class, Transport, still exists, but it now has a more specific
> role: a helper class that applies TransportDescriptions. And is renamed
> to JsepTransport as a result.
>
> TransportController is now the entity primarily responsible for managing
> TransportChannels. It also starts storing pointers to the DTLS and ICE
> chanels separately, which will make it easier to remove
> TransportChannel/TransportChannelImpl in a subsequent CL.
>
> BUG=None
>
> Committed: https://crrev.com/bd28681d02dee8c185aeb39207e8154f0ad14a37
> Cr-Commit-Position: refs/heads/master@{#15450}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2553043004
Cr-Commit-Position: refs/heads/master@{#15452}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15450}
This isn't a performance test, so it may be running in a slow
environment, and shouldn't be subject to strict timeouts.
BUG=webrtc:6801
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2539183005
Cr-Commit-Position: refs/heads/master@{#15370}
We still DCHECK for RTP, but not RTCP. RTCP packets can be sent before
offer/answer negotiation is complete, due to this bug:
https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
This bug can only occur if the RTCP mux policy is "require", which is
why we started hitting it recently (the default in unit tests was
recently changed to "require").
BUG=webrtc:6776
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2542233002
Cr-Commit-Position: refs/heads/master@{#15369}
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
