This test was only testing codec vendor functionality.
Bug: webrtc:360058654
Change-Id: I5763e766a44f6bb1542c4281b1d6c177a52c8c74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375600
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43811}
The new class "CodecVendor" is intended to handle all logic dealing
with codecs. This CL is a no-behavior-change CL, later CLs will
change the logic.
Bug: webrtc:360058654
Change-Id: I44e76f0e0bd364eeb7d4506f3e01e9e00e2843a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43806}
This allows us to verify consistency of codec lists in more places.
Bug: webrtc:360058654
Change-Id: Ibd0d10579c4b8058031db0df458e8fc9e2181152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371921
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43594}
The negotiation of encrypted header extensions has already been enabled in Chromium, https://chromium-review.googlesource.com/c/chromium/src/+/5933829. Hence, it make sense to enable the encryption of header extensions by default also in webRTC environment so that all the tests run by taking this into considiration when new changes are made.
Bug: webrtc:358039777
Change-Id: I141fac01b0eb0f2ce5a0a365736f0dcf9f21ddcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43573}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
For an offer in sendrecv direction, if for example it can send H.265
level 5.2 while receiving 6.0, setCodecPreferences on offerer's transceiver will currently remove H.265 from the offer SDP, since currently we do a precise level match on send_recv_codecs with the codecs from setCodecPreferences.
Update the matching logic to ignore the level when matching.
Bug: chromium:41480904
Change-Id: Id0f89cbf117ce62249a99257dcce18b35f407cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369960
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43477}
On a sendrecv m-line, the offered level-id represents the maximum that
can be both sent and received; on a sendonly m-line, the offered
level-id represents the maximum that can be sent; on a recvonly m-line,
the offered level-id represents the maximum that can be received.
Also according to RFC 7798 section 5, the highest level indicated by the
answer is either equal to or lower than that in the offer
Bug: chromium:41480904
Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43425}
- Avoid redundant get() when dereferencing smartpointers
- Use const ref instead of copy for RtpExtension
- Use `.empty()` instead of `.size() == 0`
- Remove some unused using declarations
Bug: None
Change-Id: I0dfdc0dfdf165f153c9ba119c115cd492e9599fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43334}
According to RFC 6904 a header extension MAY be offered both encrypted and unecrypted. In the case when encryption is enabled the encrypted version SHOULD be used and vice versa. However, this is under the assumption that both peers actually offer the same extension header both encrypted and unecrypted. With this PR we tighten the negotiation rules to the encryption option SHOULD be the same both in the sender and receiver in order to not drop the extension. Especially, see test `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionEnabled` and `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionDisabled`.
Bug: chromium:40623740
Change-Id: I68c65a776fcf7be97aaf60a797594c4361a06800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363940
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43191}
A number of unit tests assume that payload types will be assigned
without generating an offer. These are flushed out by running tests
with the --force_fieldtrials=WebRTC-PayloadTypesInTransport argument.
Bug: webrtc:360058654
Change-Id: I17cd5bfa275904a9630068190b1cd246e9ce8741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43127}
By this change we aim to remove the flag enable-webrtc-srtp-encrypted-headers.
Bug: chromium:40623740
Change-Id: I74692c90ff1caf2a11d7b73211c1ae4472edfb4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362740
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43105}
This adds payload types to the codecs at the time when offer
is being generated, if they are unassigned at that point.
Bug: webrtc:360058654
Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43033}
This provides a way to tell the SDP generator to use a specific list
of codecs, rather than trying to compute what list to send.
Preparatory to making codec decisions per-transceiver.
Bug: webrtc:42226302
Change-Id: I1b7d4e55ed7a0546394b74820b4e51434ef86ad9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349620
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42247}
These are aliases for cricket::Codec.
Also remove internal usage
Bug: b/42225532
Change-Id: I220b95260dc942368cb6280432a058159eec8700
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42194}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
since it can be enabled locally using the RTP header extensions API
Bug: webrtc:15057
Change-Id: Id15d26ab858d88769939974f2a7ae4327df925b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41650}
This adds neccessary checks for SDP negotiation with HEVC.
Test: Manually apply the CL on Chromium and enable HEVC HW encoder,
and add HEVC profiles in rtc video decoder/encoder factory, H265 is
negotiated in SDP with correct FMTP lines added.
Bug: webrtc:13485
Change-Id: I5557b20b646cc96c5acb578521204fe10df0dcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330202
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41357}
This reverts commit 1ae700a9233ed647e1b4080c0fcb48f61a0cca0a.
Reason for revert: Potential root cause of crbug.com/1504351
Original change's description:
> Make Codec::Matches also consider packetization
>
> If it's not considered it can lead to payload IDs erroneously being
> reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
>
> Bug: webrtc:15473
> Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41153}
Bug: webrtc:15473 chromium:1504351
Change-Id: I87fb671d76c3b17beb65124603cc040bb9bf4fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329201
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41285}
and unify algorithms a bit more.
BUG=webrtc:15214
Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
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@{#41244}
If it's not considered it can lead to payload IDs erroneously being
reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
Bug: webrtc:15473
Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41153}
- Adds namespaces which allows removing redundant using declarations and
static keywords.
- Adds some new using declarations and make sure they are all fully
qualified.
- Use std::tuple instead of testing::tuple.
Bug: None
Change-Id: If2ca3c0b94efc8a96da8d2dada0585370e85a7e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41151}
which is not doing anything specific to plan-b.
BUG=None
Change-Id: Ic214b10a9c3021a8ca93601453d6eb42b84f2d84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325529
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@{#41082}
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}
in order to not regress existing use-cases while following rules
described by the specification. This change now makes the existing
regression test pass after the spec-compliant modifications.
BUG=chromium:1051821
Change-Id: Ia384adf9a172ed88b5ec6a3cc5c478764a686cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299002
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@{#39726}
This reverts commit be03c0971863c9e6807fcbdb4175754e8242a652.
Causes regression in web projects that
1/ add a stopped-by-default extension in SRD
2/ call createAnswer
3/ munge the stopped-by-default extension back in SLD
4/ create a subsequent offer and expect the extension to be present
BUG=chromium:1051821
Change-Id: If77f77c2c0ac19625f502fb8a07facd151475807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298744
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39655}
as described in https://w3c.github.io/webrtc-extensions/#modifications-to-existing-procedures-0
"For each RTP header extension "e" listed in
[[HeaderExtensionsToOffer]] where direction is not "stopped", an
"a=extmap" line, as specified in [RFC5285], section 5
This avoids including them in case they are stopped on one
transceiver but not the other. Also, this allows extensions to
be removed from a subsequent offer.
See also
https://github.com/w3c/webrtc-extensions/issues/140
BUG=chromium:1051821
Change-Id: I4d7462f939ce4cd5d8c2331bc038200fe18f70e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291703
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39242}
for video dealing with both the case where there is no common media
codec as well as only a red/ulpfec/flexfec codec in common for video
and only RED/CN in common for audio
BUG=webrtc:4957,webrtc:14069
Change-Id: I1c888b4f77199aade8122051c31b690dc2fd5925
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262642
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36920}
