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}
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}
which is only used in tests.
BUG=None
Change-Id: If215ad84e6756af2ee90777a27376400f8f4d8e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294721
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@{#39450}
verifying these are transferred to the individual m-lines.
Also verify that mixed usage both at session level as well as
media level is not allowed as described in
https://www.rfc-editor.org/rfc/rfc5285#section-6
BUG=None
Change-Id: Iade387817c9f31362d0a26c5f13a3012c72b51b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39363}
This reverts commit 815522782a.
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}
increasing the line coverage to 100%.
Note that code coverage is still a very basic metric and
the tests still do not enforce the length of the attributes
nor their token-char-ness defined in
https://www.rfc-editor.org/rfc/rfc8830.html#section-2
BUG=webrtc:14745
Change-Id: If5de5e1f8c32f7dae029916c6cd0a4f2b094f672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286900
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@{#38864}
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}
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}
This is a reland of commit ad6807805d
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 ad6807805d.
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}
This reverts commit 37ee0f5e59.
Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642
Original change's description:
> Use backticks not vertical bars to denote variables in comments for /pc
>
> Bug: webrtc:12338
> Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34575}
TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34577}
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.
This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).
PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.
C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/
Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.
Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
Adds a signaling unit tests which asserts that a flexfec
offer negotiates flexfec in the answer
BUG=webrtc:8151
Change-Id: Ica6bfe2bdde1a035cdd429d636fefb7f751062d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204680
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33816}
regression from
https://webrtc-review.googlesource.com/c/src/+/197813
which attempted to cast the unsupported content type with
a sctp protocol to a application/datachannel one.
BUG=chromium:1171965
Change-Id: I87c63da83b9f49d968e9b045bb1079f687ab226e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#33100}
The extmap-allow-mixed SDP attribute signals that one- and two-byte RTP
header extensions can be mixed. In practice, this also means that WebRTC
will support two-byte RTP header extensions when this is signaled by
both peers.
Bug: webrtc:9985
Change-Id: I80a3f97bab162c7d9a5acf2cae07b977641c039d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197943
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33036}
This is a reland of 239f92ecf7
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
This reverts commit 239f92ecf7.
Reason for revert: Breaks downstream projects.
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
TBR=kthelgason@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com
Change-Id: I055fe001fe2757d79be7c304eccc43a8e3104f69
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32411}
This carries around unsupported content descriptions
(i.e. things where webrtc does not understand the media type
or protocol) in a special data type so that a rejected content or
mediasection is added to the answer SDP.
BUG=webrtc:3513
Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32410}
This reverts commit 20b701f3d7.
Reason for reland: Reverting did not affect the test regression.
Original change's description:
> Revert "sdp: parse and serialize b=TIAS"
>
> This reverts commit c6801d4522.
>
> Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
>
> One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
>
> Original change's description:
> > sdp: parse and serialize b=TIAS
> >
> > BUG=webrtc:5788
> >
> > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31729}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:5788
> Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31762}
TBR=nisse@webrtc.org
Bug: webrtc:5788
Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31816}
This reverts commit c6801d4522.
Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
Original change's description:
> sdp: parse and serialize b=TIAS
>
> BUG=webrtc:5788
>
> Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31729}
TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:5788
Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31762}
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}
It was used only to provide parameters for MediaTransport.
Bug: webrtc:9719
Change-Id: I42e451ef84251ecf2b15010c7a3923b6fa2436be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177350
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31541}
Our implementation accepts TCP candidates with a missing tcptype
field, treating this as a passive candidate.
However, if you try to convert such a candidate to SDP and back,
which chromium started to do at some point, this was resulting in an
error. This CL fixes that.
Bug: webrtc:11423
Change-Id: Iec48d340f421f63f2b7a16c9496ea92ccd165981
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31026}
