It's possible that a peer can signal the same payload with multiple
packetization options. As such, we shouldn't try to fall back to default
packetization until we have considered all the alternatives.
Bug: webrtc:15473
Change-Id: I21772b4d8c53819d1c3105988551ebdbea0df045
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320241
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40775}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
This is a reland of commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I123d1134a212f65cfbc90ecec9013d0aafebd9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308721
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40294}
This reverts commit 49ace8b6548cda6d4ba74abfca9b616f56dbf9bc.
Reason for revert: Breaks downstream projects
Original change's description:
> Merge the codec types
>
> This allows simplifying code in the codebase to be able to remove a lot
> of templated code and special casing for either AudioCodec and VideoCodec.
> Code simplifications will come in later changes.
>
> Bug: webrtc:15214
> Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40276}
Bug: webrtc:15214
Change-Id: I57778cccc3a13eb9f955f6ece054dee0ff5a7e92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40278}
This allows simplifying code in the codebase to be able to remove a lot
of templated code and special casing for either AudioCodec and VideoCodec.
Code simplifications will come in later changes.
Bug: webrtc:15214
Change-Id: I6e75e6ea725163feb6cc4eb49f37b4722d6c6689
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308501
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40276}
The preferred method to create codecs is to use the function
cricket::CreateAudioCodec or cricketCreateVideoCodec.
Empty codec objects are deprecated and should be replaced
with alternatives such as methods returning an
absl::optional object instead.
Bug: webrtc:15214
Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40226}
Part 1 of the migration towards merging the types.
Any method that could belong to the Codec type was moved, the others
are deprecated.
Alternatives to the AudioCodec and VideoCodec constructors are introduced
to allow creating objects of an indefinite type without having to
reference the old classes.
Bug: webrtc:15214
Change-Id: I20e1aa32962821cad98e9a92c2ec86f8f75e5dd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40213}
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.
Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
some codecs like RED and telephone-event have fmtp lines which
do not conform to the list-of-key=value convention. Add support
for parsing and serializing this by setting the name to the empty
string.
BUG=webrtc:11640
Change-Id: Ie3ef7c98f756940f97d27a39af0574aa37949f74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31609}
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}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
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}
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}
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}
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}
(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}
SetCodecPreferences allows clients to filter and reorder codecs in their
SDP offer and answer.
Bug: webrtc:9777
Change-Id: I716bed9b06496629b45210883b286f599c875239
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129727
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27817}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
Bug: webrtc:9419
Change-Id: I12ef6f85ccef7dae3afe9ecff99725af13d551e2
Reviewed-on: https://webrtc-review.googlesource.com/c/106684
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25246}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and
SocketAddress are kept behind gtest's #ifdef UNIT_TEST.
Bug: webrtc:8982
Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70
Reviewed-on: https://webrtc-review.googlesource.com/64143
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22916}
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}
Unfortunately, H264 makes it non-trivial to compare video formats for
equality. For every video format besides H264 it's enough to look at the
name, but for H264, we need to dig into the parameters. This logic is
currently in several places, and this CL unifies it to one place.
Bug: webrtc:7925
Change-Id: I83a516b108d6b4d6792fd0bf1d24296916d9f7fa
Reviewed-on: https://webrtc-review.googlesource.com/25120
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20853}
We will have to convert from webrtc::SdpVideoFormat to
cricket::VideoCodec in a couple of places until
cricket::WebRtcVideoEncoderFactory is gone. It will be convenient to
have the conversion logic in a common place.
Bug: webrtc:7925
Change-Id: Ie5e88599f28aeea647e936300c04f9071daffd53
Reviewed-on: https://webrtc-review.googlesource.com/4840
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20048}
We currently suppress warnings for bad constructs in media/. Still, the
warnings are causing problems when trying to include header files from
this directory. This CL cleans up some of the bad constructs.
Bug: None
Change-Id: I808ad39eb23870d20fa5bb05429b50c9078543ae
Reviewed-on: https://webrtc-review.googlesource.com/4541
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20039}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
