* changing the defaults to unified-plan,
* expect unified-plan by default
* only include the msid-semantic line in answers when the
offer contained a sign of msid being used
BUG=webrtc:10421
Change-Id: I79364c2dd801d37adea119375e4dc01edf55b3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331800
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41480}
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
by also taking into account any a=msid: line in addition to
msid-semantic. Also document issues with msid-semantic generation and unify support determination by removing the msid_supported flag.
BUG=webrtc:10421
Change-Id: Icea554ebd1998f2b526846457029eff6854a772a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41386}
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
as preparation for H265 work.
BUG=webrtc:15703
Change-Id: Ib6e0afa5ccbb8172a70d4e4eb876639559070fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329981
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41350}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
allowing for a lot of de-templating
BUG=webrtc:15214
Change-Id: Ibe1a5f5d704564566f24c496822a4308ba23c4dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319160
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40774}
which is not a generic SDP serializer but only deals with the
simulcast SDP.
BUG=None
Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40761}
move this a bit later in the process since the current handling will consider two ssrc-lines with a cname in the same RTX FID ssrc-group to be part of separate streams due to the different randomly assigned msids. This leads to a misdetection as plan-b SDP.
BUG=None
Change-Id: Ie8acce9c2c7fb9eabda479b90e8cc7406dcb1696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40701}
remove some of the templating around the Codec-derived types and
use more modern C++ loops.
BUG=webrtc:15214
Change-Id: I2710628741deca647e7ae88f5966ec7c7f12669a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311260
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40475}
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}
while not really covered by
https://www.rfc-editor.org/rfc/rfc5576.html#section-4.2
and using the same SSRC for RTX and primary payload may work
since payload type demuxing *could* be used is not a good idea.
This also applies to flexfec's FEC-FR.
For the nonstandard SIM ssrc-group duplicates make no sense.
This rejects duplicates for unknown ssrc-groups as well.
BUG=chromium:1454860
Change-Id: I3e86101dbd5d6c4099f2fdb7b4a52d5cd0809c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308820
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40292}
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}
This reverts commit 815522782a92e168b80edc760b2e53e4d0e4ea0d.
Reason for revert: Breaks a downstream project.
The internal investigation is still in-progress.
Original change's description:
> sdp: add rtcp-fb:* lines for common feedback
>
> which potentially allows switching to that pattern in the future.
> Video FEC mechanisms (ulpfec, flexfec-03, RED) that currently
> do not have any feedback parameters but will still be considered "common" and feedback may be sent for them.
>
> For audio this causes rtcp-feedback to be sent for G711 and G722 if negotiated.
>
> BUG=webrtc:14802
>
> Change-Id: I54852d39e176f918d4b36462526ceb40617b8fbe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290702
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39224}
Bug: webrtc:14802
Change-Id: I4dc3c0c53ad1bc06050c0d73b088303312ac58b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293020
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39296}
In current state, the SDP parser in webrtc is not backward compatible with clients that might still be using RTP data channels.
Obviously, this isn't there is no such usecase in webrtc since the code is deleted, but in Meta we still use it and would like
to be able to negotiate between clients that offer RTP data channels.
Instead of erroring the parsing procedure, we can parse it as unsupported media in the client that no longer supports RTP data channels.
Replaced the existing test that expects parsing failures with a test that validates that the content was parsed as unsupported media.
Bug: webrtc:14872
Change-Id: I4c105cf55e33b8c19b2849e16148b8175053c40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291190
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39252}
which potentially allows switching to that pattern in the future.
Video FEC mechanisms (ulpfec, flexfec-03, RED) that currently
do not have any feedback parameters but will still be considered "common" and feedback may be sent for them.
For audio this causes rtcp-feedback to be sent for G711 and G722 if negotiated.
BUG=webrtc:14802
Change-Id: I54852d39e176f918d4b36462526ceb40617b8fbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290702
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39224}
preparing to put them at session level when max-bundle is set.
Drive-by: move m= serialization to helper.
BUG=None
Change-Id: I04d918ee8eb70c0cc40baf8ebc12054c6b3a2a15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288820
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38950}
https://www.rfc-editor.org/rfc/rfc8830.html#section-3.2.2
says
Check if a MediaStream with the same WebIDL "id"
attribute already exists. If not, create it.
Ignoring duplicates here satisfies this and brings the behavior
closer to Firefox:
https://github.com/w3c/webrtc-pc/issues/2803
Also make tests use a std::string for the sdp input string.
BUG=webrtc:14745
Change-Id: Iccaabc08d865b779416f6ba4d2dfd5cff04133f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286422
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38840}
parse
a=msid:<stream_id>
since JSEP stipulates sending this syntax as track identifers
have become meaningless. The track id will be set to a random string.
a=msid:<stream_id> <track_id>
remains supported for backward compability.
BUG=webrtc:14729
Change-Id: I86c073eb97cd613324271125de18a773235fc79d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38814}
The DCHECK crashes debug builds running some applications such as Webex.
Bug: None
Change-Id: I0061286c4c1d04964678a00014896f1fccd4685d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276460
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38644}
to allow for downstream users to upgrade.
BUG=chromium:1338902
Change-Id: Ie1205ad2c9c1be3f4ed8e133b1a5e54afd04ebd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268193
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37501}
to allow for downstream users to upgrade.
BUG=chromium:1338902
Change-Id: If6b56ab63f7859c13e9ebc70326e1088e5dfff1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268141
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37475}
RID is defined for multiple usages in RFC 8851, but we only support
usage with a=simulcast as specified in RFC 8853.
Bug: chromium:1341043
Change-Id: Ie72074c5b394bdc41865938a86ec9c7629e1f5e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267628
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37417}
This is a reland of commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: webrtc:14140
Change-Id: I63a37aacea6b9e0a9d7570b8422849275eb69aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264544
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37066}
This reverts commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5.
Reason for revert: Speculative revert due to consistent Mac browser
test failures preventing WebRTC from rolling int Chromium:
https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/10410/overview
"Failed to parse SessionDescription. a=rtpmap:103 ISAC/16000 Duplicate payload type with conflicting codec name, clock rate or number of channels."
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: None
Change-Id: Ic9c06c9309bb68bd94bfdd2e30ffd6ff96f6812b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37064}
since something like
rtpmap:96 VP8/90000
rtpmap:96 VP9/90000
or
rtpmap:97 ISAC/32000
rtpmap:97 ISAC/16000
is wrong. Note that fmtp or rtcp-fb are not taken into account.
Also note that sending invalid static payload types now throws an error.
Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
BUG=None
Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37028}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35440}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
