GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".
If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).
BUG=webrtc:5222, 628400
Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.
This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
Added initial support for MediaSessionDescriptionFactory to pick different codecs based on communications direction (sendrecv, sendonly, recvonly, inactive) specifically for audio.
This adds some more degradation options for the answer: depending on answer options, it's now possible to degrade to INACTIVE from any offer, as well as to either RECVONLY or SENDONLY from a SENDRECV offer.
The set of "codecs" used for testing the answer was compiled using this spreadsheet:
https://docs.google.com/a/google.com/spreadsheets/d/1nVIfZLsFo5YK10_e80BCAADZnnRQ1devwwwAGmqJPow/edit?usp=sharing
I should probably condense it into a smaller table and put in the source.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1956343002
Cr-Commit-Position: refs/heads/master@{#13126}
This fixes a client breakage by adding back the RtpHeaderExtension temporarily
so that it can be fixed in the client before being removed in webrtc.
BUG=
CQ_INCLUDE_TRYBOTS=tryserver.chromium.linux:linux_chromium_rel_ng;tryserver.chromium.win:win_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2024153002
Cr-Commit-Position: refs/heads/master@{#12977}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
Some tests were passing in a local description created from hard-coded
SDP strings, which won't work in the future (since some attributes such
as the fingerprint and ICE ufrag/pwd are non-modifiable). These tests
now do the typical approach of calling CreateOffer and modifying the
result if necessary.
Also added some non-const versions of the SessionDescription accessor
helper functions, since that makes it much easier to modify a
SessionDescription. Previous alternatives were re-implementing the
helper methods from scratch, or converting the description to SDP,
modifying it, and converting it back.
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1966333002 .
Cr-Commit-Position: refs/heads/master@{#12704}
The CNAME is generated in the PeerConnection constructor and is populated through the MediaSessionOptions.
A default cname will be set in the MediaSessionOptions constructor.
BUG=webrtc:3431
Review-Url: https://codereview.webrtc.org/1871993002
Cr-Commit-Position: refs/heads/master@{#12650}
This field only existed as an implementation detail for getting the
codecs sorted, so it doesn't need to be in the public interface.
It cluttered the code and undesirably affected codec comparisons,
causing the video encoder to be reconfigured if a codec's preference
changed but nothing else did.
BUG=webrtc:5690
Review URL: https://codereview.webrtc.org/1845673002
Cr-Commit-Position: refs/heads/master@{#12349}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
RFC 5245 allows an ICE restart to occur on only one media section.
However, before this CL, if an endpoint attempted to do this, we would
change our local ICE ufrag/pwd in every media section.
Also did some refactoring, turning the transport options from
mediasesion.h into a map.
Review URL: https://codereview.webrtc.org/1671173002
Cr-Commit-Position: refs/heads/master@{#11728}
