It was defined unconditionally and the code for non-HAVE_SRTP was unmaintained
and failed to compile.
BUG=webrtc:7294
Review-Url: https://codereview.webrtc.org/2729373002
Cr-Commit-Position: refs/heads/master@{#17074}
This change adds a flag, use_sctpmap, to DataContentDescription. The deserialization code sets the flag based on the format of the m= line.
There were already unit tests using SDP in the new format, so I just updated them to check use_sctpmap was set as expected.
The change to mediasession copies use_sctpmap from the offered DataContentDescription to the answer.
I haven't figured out how to test this change yet, but wanted to get feedback before continuing.
BUG=chromium:686212
Review-Url: https://codereview.webrtc.org/2690943011
Cr-Commit-Position: refs/heads/master@{#16686}
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.
In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using
git grep -l ' ASSERT(' | grep -v common.h | \
xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".
BUG=None
Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
These functions currently copy cricket::Codec classes by value which is
expensive since they contain e.g. std::map<std::string, std::string>
containers with parameters. This CL avoids copying them altogether.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2493733003
Cr-Commit-Position: refs/heads/master@{#15040}
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.
This cl require some upstream projects to change first.
BUG=webrtc:5332
Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
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}
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}
Reason for revert:
Breaks user code. Said code needs to stop using scoped_ptr!
Original issue's description:
> Remove webrtc/base/scoped_ptr.h
>
> BUG=webrtc:5520
>
> NOTRY=True
>
> Committed: https://crrev.com/65fc62e9dd8a8716db625aaef76ab92f542ecc5a
> Cr-Commit-Position: refs/heads/master@{#12684}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review-Url: https://codereview.webrtc.org/1965063003
Cr-Commit-Position: refs/heads/master@{#12686}
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 propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
Reason for revert:
Broke the Chromium build by introducing static initializers.
Original issue's description:
> Accept all the media profiles required by JSEP.
>
> JSEP section 5.1.3 states that:
> Any profile matching the following patterns MUST be accepted:
> "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
>
> NOTRY=True
> BUG=webrtc:5638
>
> Committed: https://crrev.com/b7f425ab68ec58e2a5beaaf5ef79f50f1982c6f9
> Cr-Commit-Position: refs/heads/master@{#12338}
TBR=deadbeef@webrtc.org,pthatcher@webrtc.org,avi@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5638
Review URL: https://codereview.webrtc.org/1882923002
Cr-Commit-Position: refs/heads/master@{#12351}
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}
JSEP section 5.1.3 states that:
Any profile matching the following patterns MUST be accepted:
"RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"
NOTRY=True
BUG=webrtc:5638
Review URL: https://codereview.webrtc.org/1880913002
Cr-Commit-Position: refs/heads/master@{#12338}
This fixes a couple major issues.
#1: If the payload type that an RTX codec refers to has been reassigned, and then the RTX codec is added in a subsequent offer, it refers to the wrong payload type.
#2: If we receive an offer with two payload types referring to the same codec (which we support), our answer contains both (instead of just one), which causes issues down the road since the video engine only supports one payload type per codec.
BUG=webrtc:5450,webrtc:5499
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1616033002 .
Cr-Commit-Position: refs/heads/master@{#11880}
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}
