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Author SHA1 Message Date
Florent Castelli
8038541a4f Update the header extensions capabilities with mid, rid and rrid 
Video and audio senders are missing mid, rid and rrid extensions in
their GetCapabilities call.

Bug: chromium:1007894
Change-Id: Ie9edba28ae32fda5e501913cac694f43bfb185ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156560
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29493}
 2019-10-15 14:45:58 +00:00
Bjorn A Mellem
8e1343aeda Add an alt-protocol to SDP to indicate which m= sections use a plugin transport. 
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport.  When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section.  This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section.  Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.

This change adds an a=x-alt-protocol: line to SDP.  The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line.  This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.

Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data.  It is
still not possible to use it for audio but not video, or vice versa.

PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media.  It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels.  PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.

JsepTransport now negotiates use of the datagram transport independently for
media and data channels.  It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.

Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
 2019-09-30 23:10:34 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Jonas Olsson
a4d873786f Format almost everything. 
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
 2019-07-08 13:45:15 +00:00
Harald Alvestrand
c5effc2453 Remove DataContentDescription shim 
Bug: webrtc:10597
Change-Id: Id0cbb78846d2b248bc2ab650eb7c06b50bc825bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28275}
 2019-06-13 18:33:40 +00:00
Bjorn A Mellem
c85ebbe766 Reland: Implement true negotiation for DatagramTransport with fallback to RTP. 
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport.  If the answerer supports datagram transport, it will
parse this line and create a datagram transport.  It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).

If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport.  If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.

Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto.  Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP.  This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.

Negotiation consists of four parts:
 1. DatagramTransport exposes transport parameters for both client and server
 perspectives.  The client just echoes what it received from the server (modulo
 any fields it might not have understood).

 2. SDP adds a x-opaque line for opaque transport parameters.  Identical to
 x-mt, but this is specific to datagram transport and goes in each m= section,
 and appears in the answer as well as the offer.
  - This is propagated to Jsep as part of the TransportDescription.
  - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
    media_session.cc, webrtc_sdp.cc

 3. JsepTransport/Controller:
  - Exposes opaque parameters for each mid (m= section).  On offerer, this means
    pre-allocating a datagram transport and getting its parameters.  On the
    answerer, this means echoing the offerer's parameters.
  - Uses a composite RTP transport to receive from either default RTP or
    datagram transport until both offer and answer arrive.
  - If a provisional answer arrives, sets the composite to send on the
    provisionally selected transport.
  - Once both offer and answer are set, deletes the unneeded transports and
    keeps whichever transport is selected.

 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.

Bug: webrtc:9719
Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28198}
 2019-06-07 20:14:36 +00:00
Bjorn Mellem
7e8de0bf2d Revert "Implement true negotiation for DatagramTransport with fallback to RTP." 
This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.

Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.

Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
> 
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport.  If the answerer supports datagram transport, it will
> parse this line and create a datagram transport.  It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
> 
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport.  If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
> 
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto.  Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP.  This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
> 
> Negotiation consists of four parts:
>  1. DatagramTransport exposes transport parameters for both client and server
>  perspectives.  The client just echoes what it received from the server (modulo
>  any fields it might not have understood).
> 
>  2. SDP adds a x-opaque line for opaque transport parameters.  Identical to
>  x-mt, but this is specific to datagram transport and goes in each m= section,
>  and appears in the answer as well as the offer.
>   - This is propagated to Jsep as part of the TransportDescription.
>   - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
>     media_session.cc, webrtc_sdp.cc
> 
>  3. JsepTransport/Controller:
>   - Exposes opaque parameters for each mid (m= section).  On offerer, this means
>     pre-allocating a datagram transport and getting its parameters.  On the
>     answerer, this means echoing the offerer's parameters.
>   - Uses a composite RTP transport to receive from either default RTP or
>     datagram transport until both offer and answer arrive.
>   - If a provisional answer arrives, sets the composite to send on the
>     provisionally selected transport.
>   - Once both offer and answer are set, deletes the unneeded transports and
>     keeps whichever transport is selected.
> 
>  4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
> 
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}

TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org

Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
 2019-06-07 06:17:50 +00:00
Bjorn A Mellem
71c6482baf Implement true negotiation for DatagramTransport with fallback to RTP. 
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport.  If the answerer supports datagram transport, it will
parse this line and create a datagram transport.  It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).

If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport.  If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.

Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto.  Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP.  This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.

Negotiation consists of four parts:
 1. DatagramTransport exposes transport parameters for both client and server
 perspectives.  The client just echoes what it received from the server (modulo
 any fields it might not have understood).

 2. SDP adds a x-opaque line for opaque transport parameters.  Identical to
 x-mt, but this is specific to datagram transport and goes in each m= section,
 and appears in the answer as well as the offer.
  - This is propagated to Jsep as part of the TransportDescription.
  - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
    media_session.cc, webrtc_sdp.cc

 3. JsepTransport/Controller:
  - Exposes opaque parameters for each mid (m= section).  On offerer, this means
    pre-allocating a datagram transport and getting its parameters.  On the
    answerer, this means echoing the offerer's parameters.
  - Uses a composite RTP transport to receive from either default RTP or
    datagram transport until both offer and answer arrive.
  - If a provisional answer arrives, sets the composite to send on the
    provisionally selected transport.
  - Once both offer and answer are set, deletes the unneeded transports and
    keeps whichever transport is selected.

 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.

Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
 2019-06-07 01:09:04 +00:00
Harald Alvestrand
1716d39714 Let SessionDescription take ownership of MediaDescription 
This documents in the API what is already true in the
implementation - that SessionDescription will eventually
delete MediaDescription objects passed to it.

The old API is preserved for backwards compatibility, but
marked as RTC_DEPRECATED.

Bug: webrtc:10701
Change-Id: I9a822b20cf3e58c5945fa51dbf6082960a332de8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28144}
 2019-06-03 20:07:37 +00:00
Guido Urdaneta
cecf87ff29 Reland "Change default secure SCTP protocol to UDP/DTLS/SCTP" 
This reverts commit 4436887ed2d3324279e0f2e091c9e9355392721a.

Reason for revert: The original revert was intended to be temporary.

Original change's description:
> Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP"
> 
> This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.
> 
> Reason for revert: Will temporarily revert to fix an issue and reland afterwards.
> 
> Original change's description:
> > Change default secure SCTP protocol to UDP/DTLS/SCTP
> > 
> > The old value - DTLS/SCTP - is not standards conformant,
> > and the new value should be parsable since Chrome M61.
> > 
> > Bug: webrtc:7706
> > Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27940}
> 
> TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:7706
> Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28121}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org,guidou@webrtc.org

Change-Id: I381fa18b644874c20ddaa4cd13fec79a5fd9555a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7706
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139246
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28122}
 2019-05-31 11:08:03 +00:00
Guido Urdaneta
4436887ed2 Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP" 
This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.

Reason for revert: Will temporarily revert to fix an issue and reland afterwards.

Original change's description:
> Change default secure SCTP protocol to UDP/DTLS/SCTP
> 
> The old value - DTLS/SCTP - is not standards conformant,
> and the new value should be parsable since Chrome M61.
> 
> Bug: webrtc:7706
> Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27940}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:7706
Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28121}
 2019-05-31 10:16:10 +00:00
Harald Alvestrand
8d3d6cf908 SCTP: Treat message size zero as "responder selects" 
This also refactors some of the code in peerconnection for
handling SCTP transports to be internal to the webrtc::SctpTransport
class, rather than being in peerconnection.

Bug: webrtc:10358, webrtc:10629
Change-Id: I15ecf95c199f56b08909e5a9311d446a412ed162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137041
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27960}
 2019-05-16 11:03:17 +00:00
Harald Alvestrand
c3f4820e12 Change default secure SCTP protocol to UDP/DTLS/SCTP 
The old value - DTLS/SCTP - is not standards conformant,
and the new value should be parsable since Chrome M61.

Bug: webrtc:7706
Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27940}
 2019-05-14 18:36:47 +00:00
Harald Alvestrand
5fc28b11a0 Reland "Reland "Version 2 "Refactoring DataContentDescription class""" 
This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1.

Reason for revert: Tightened protocol name handling.

Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
> >
> > Original change's description:
> > > Version 2 "Refactoring DataContentDescription class"
> > >
> > > (substantial changes since version 1)
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > > and cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Due to usage of internal interfaces by consumers, shimming the old
> > > DataContentDescription API is needed.
> > >
> > > A new cricket::DataContentDescription class is defined, which is
> > > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > > It exposes as little functionality as possible, but supports the
> > > concerned consumer's usage
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > >

Bug: webrtc:10358
Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27933}
 2019-05-13 18:37:47 +00:00
Steve Anton
46afbf9481 Revert "Reland "Version 2 "Refactoring DataContentDescription class""" 
This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.

Reason for revert: fuzzer failures

Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
>
> Original change's description:
> > Version 2 "Refactoring DataContentDescription class"
> >
> > (substantial changes since version 1)
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > and cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Due to usage of internal interfaces by consumers, shimming the old
> > DataContentDescription API is needed.
> >
> > A new cricket::DataContentDescription class is defined, which is
> > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > It exposes as little functionality as possible, but supports the
> > concerned consumer's usage
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> >
> > Bug: webrtc:10358
> > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27853}
>
> Bug: webrtc:10358
> Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27896}

TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org

Change-Id: Ied6d9fb96aafe9c957f2658b34b5331b1f359b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135986
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27917}
 2019-05-10 18:16:09 +00:00
Harald Alvestrand
37f2b43274 Reland "Version 2 "Refactoring DataContentDescription class"" 
This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c

Original change's description:
> Version 2 "Refactoring DataContentDescription class"
> 
> (substantial changes since version 1)
> 
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
> 
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
> 
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
> 
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
> 
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> 
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> 
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}

Bug: webrtc:10358
Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27896}
 2019-05-09 18:15:48 +00:00
Harald Alvestrand
141c0ad8ab Revert "Version 2 "Refactoring DataContentDescription class"" 
This reverts commit 14b2758726879d21671a21291dfed8fb4fd5c21c.

Reason for revert: Internal import failed.

Original change's description:
> Version 2 "Refactoring DataContentDescription class"
> 
> (substantial changes since version 1)
> 
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
> 
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
> 
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
> 
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
> 
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> 
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> 
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}

TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org

Change-Id: Ibc16ba14c1cbf50345a9b79151b79df140482539
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27855}
 2019-05-05 19:00:13 +00:00
Harald Alvestrand
14b2758726 Version 2 "Refactoring DataContentDescription class" 
(substantial changes since version 1)

This CL splits the cricket::DataContentDescription class into
two classes: cricket::RtpDataContentDescription (used for RTP data)
and cricket::SctpDataContentDescription (used for SCTP only).

SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".

Due to usage of internal interfaces by consumers, shimming the old
DataContentDescription API is needed.

A new cricket::DataContentDescription class is defined, which is
a shim over RtpDataContentDescription and SctpDataContentDescription.
It exposes as little functionality as possible, but supports the
concerned consumer's usage

Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#

Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700

Bug: webrtc:10358
Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27853}
 2019-05-05 13:22:21 +00:00
Danil Chapovalov
c6d1d24de8 Revert "Reland "Refactoring DataContentDescription class"" 
This reverts commit 26bf7c4682c7ec72465a1d4d6485d2ec01f671cc.

Reason for revert: breaks downstream test

Original change's description:
> Reland "Refactoring DataContentDescription class"
> 
> This reverts commit 1859dc04fd8bd35a3d2ee1140bde3eac210bb0c2.
> 
> Reason for revert: Issue likely unrelated to this CL.
> 
> Original change's description:
> > Revert "Refactoring DataContentDescription class"
> >
> > This reverts commit 8a9193c217d818fea77b9540bd4ca7ebad53db76.
> >
> > Reason for revert: Breaks downstreams
> >
> > Original change's description:
> > > Refactoring DataContentDescription class
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::DataContentDescription (used for RTP data) and
> > > cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Bug: webrtc:10358
> > > Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#27651}
> >
> > TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
> >
> > Change-Id: I3b8a68cd481c41ce30eeb5ffbc5da735a9659019
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:10358
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133360
> > Reviewed-by: Seth Hampson <shampson@webrtc.org>
> > Commit-Queue: Seth Hampson <shampson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27652}
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10358
> Change-Id: Ie58f862f8c55d2a994eaee1caa107ef701b0770f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133624
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27698}

TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org

Change-Id: Ib17939d5f1e8c57652dcb34d94866654192379bb
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133880
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27702}
 2019-04-23 09:48:59 +00:00
Harald Alvestrand
26bf7c4682 Reland "Refactoring DataContentDescription class" 
This reverts commit 1859dc04fd8bd35a3d2ee1140bde3eac210bb0c2.

Reason for revert: Issue likely unrelated to this CL.

Original change's description:
> Revert "Refactoring DataContentDescription class"
>
> This reverts commit 8a9193c217d818fea77b9540bd4ca7ebad53db76.
>
> Reason for revert: Breaks downstreams
>
> Original change's description:
> > Refactoring DataContentDescription class
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::DataContentDescription (used for RTP data) and
> > cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Bug: webrtc:10358
> > Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27651}
>
> TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: I3b8a68cd481c41ce30eeb5ffbc5da735a9659019
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10358
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133360
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27652}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10358
Change-Id: Ie58f862f8c55d2a994eaee1caa107ef701b0770f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133624
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27698}
 2019-04-23 09:08:07 +00:00
Seth Hampson
1859dc04fd Revert "Refactoring DataContentDescription class" 
This reverts commit 8a9193c217d818fea77b9540bd4ca7ebad53db76.

Reason for revert: Breaks downstreams

Original change's description:
> Refactoring DataContentDescription class
> 
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::DataContentDescription (used for RTP data) and
> cricket::SctpDataContentDescription (used for SCTP only).
> 
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
> 
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> 
> Bug: webrtc:10358
> Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27651}

TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I3b8a68cd481c41ce30eeb5ffbc5da735a9659019
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133360
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27652}
 2019-04-16 20:56:06 +00:00
Harald Alvestrand
8a9193c217 Refactoring DataContentDescription class 
This CL splits the cricket::DataContentDescription class into
two classes: cricket::DataContentDescription (used for RTP data) and
cricket::SctpDataContentDescription (used for SCTP only).

SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".

Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#

Bug: webrtc:10358
Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27651}
 2019-04-16 20:34:34 +00:00
Harald Alvestrand
4d7160e41d Code cleanup: Make JsepSessionDescription::Initialize take std::unique_ptr 
Bug: none
Change-Id: I5e03ff6a6f16bd2e67fa4830e218b510b702d1d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27577}
 2019-04-12 05:42:46 +00:00
Mirko Bonadei
6a489f22c7 Fully qualify googletest symbols. 
Semi-automatically created with:

git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format

After this, two .cc files failed to compile and I have fixed them
manually.

Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
 2019-04-09 17:18:20 +00:00
Johannes Kron
8cc711a7e1 Update URI of TransportSequenceNumberV2 
The previous URI was a placeholder and is not valid. The URI
https://webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02/
must be used instead.

Bug: webrtc:10264
Change-Id: Ibabde599b5bbd116c1c5e86ba0c9c64019bf7026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27051}
 2019-03-11 10:15:52 +00:00
Johannes Kron
ce8e8677df Add support for TransportSequenceNumberV2 in SDP negotiation 
TransportSequenceNumberV2 is an experimental feature that should
not be part of the default offer. However, if we receive an offer
with this extension we should respond that we support it.

Bug: webrtc:10264
Change-Id: Id2424d421361e5d71f3a608cb8f74b63645c264a
Reviewed-on: https://webrtc-review.googlesource.com/c/123783
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26817}
 2019-02-22 12:47:04 +00:00
Elad Alon
157540ac05 Stop hard-coding default IDs for RTP extensions 
Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.

Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).

Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
 2019-02-09 01:04:35 +00:00
Mirko Bonadei
c84f661b10 Stop using Googletest legacy APIs. 
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.

This CL moves WebRTC to the new set of APIs.

More info in [1].

This CL has been generated with this script:

declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
  git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format

[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature

Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
 2019-01-31 13:23:33 +00:00
Mirko Bonadei
649a4c2ea3 [clang-tidy] Apply performance-inefficient-vector-operation fixes. 
This CL applies clang-tidy's performance-inefficient-vector-operation
[1] on the WebRTC codebase.

All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.

[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-inefficient-vector-operation.html

Bug: webrtc:10252
Change-Id: I824caab2a5746036852e00d714b89aa5ec030ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/120052
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26442}
 2019-01-29 09:45:21 +00:00
Steve Anton
64b626b03f Use Abseil container algorithms in pc/ 
Bug: None
Change-Id: If784461b54d95bdc6f8a7d4e5d1bbfa52d1a390e
Reviewed-on: https://webrtc-review.googlesource.com/c/119862
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26433}
 2019-01-29 02:33:50 +00:00
Amit Hilbuch
b7446ed257 Removing receive RIDs and Simulcast Layers. 
In the January 22nd 2019 WebRTC meeting it was agreed that an offer
for sending (or receiving) simulcast should only contain the RIDs
of the layers that are sent by the client.
This change removes the complexity that was added to support sending
and receiving the single layer (and RID) that are sent from the server.

Bug: webrtc:10076
Change-Id: I8bae1336d5cb8ba2f91c5b62332dc69e67ddfd47
Reviewed-on: https://webrtc-review.googlesource.com/c/120242
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26432}
 2019-01-29 00:54:26 +00:00
Amit Hilbuch
bcd39d483d Creating Simulcast offer and answer in Peer Connection. 
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.

Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}
 2019-01-28 18:56:02 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
 2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files 
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
 2019-01-11 17:05:20 +00:00