15 Commits

Author SHA1 Message Date
deadbeef
3e8016e1d5 Ignore "b=AS:-1" instead of treating as a hard error.
Follow up to https://codereview.webrtc.org/2989243002/.

It turns out that "b=AS:-1" was being used to mean "no bandwidth limit",
even though just omitting "b=AS" completely will do that. So we should
treat this as a soft error for now, and give applications time to
transition to doing the standard thing.

BUG=chromium:675361

Review-Url: https://codereview.webrtc.org/2995463002
Cr-Commit-Position: refs/heads/master@{#19244}
2017-08-04 00:49:30 +00:00
deadbeef
bc88c6ba98 Reject negative values for "b=AS".
It doesn't make sense to have a negative RTP session bandwidth; RFC3550
doesn't define any meaning for this. So just treat it as invalid SDP.

BUG=chromium:675361

Review-Url: https://codereview.webrtc.org/2989243002
Cr-Commit-Position: refs/heads/master@{#19221}
2017-08-02 18:26:34 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
2017-07-01 14:48:18 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
2017-06-30 20:42:44 +00:00
jbauch
5869f50f7a Support encrypted RTP extensions (RFC 6904)
Can be enabled by setting "enable_encrypted_rtp_header_extensions" in
"crypto_options" of "PeerConnectionFactoryInterface::Options" and will
only be used if both peers support it.

BUG=webrtc:3411

Review-Url: https://codereview.webrtc.org/2761143002
Cr-Commit-Position: refs/heads/master@{#18842}
2017-06-29 19:31:36 +00:00
eladalon
f184138a5f s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine
WebRtcVideoChannel and and WebRtcVideoEngine seem to have been removed, and only WebRtcVideoChannel2 and WebRtcVideoEngine2 remain, which removes the need for the "2" postfix.

BUG=None

Review-Url: https://codereview.webrtc.org/2932073002
Cr-Commit-Position: refs/heads/master@{#18531}
2017-06-12 08:16:46 +00:00
ehmaldonado
121cabbaa6 Fix webrtcsdp_unittest.
The test contained an invalid IPv6 address. It should have ":" instead of "::" as separation.

BUG=webrtc:7565

Review-Url: https://codereview.webrtc.org/2868453002
Cr-Commit-Position: refs/heads/master@{#18035}
2017-05-05 19:04:36 +00:00
zhihuang
38989e593c Parse the connection data in SDP (c= line).
Extract the remote addresses from SDP c= line on both session level and
media level. The media level address will overwrite the session level one if
exists.

WebRTC is not using c= and this is used for new SDP parsing API.

BUG=webrtc:7311

Review-Url: https://codereview.webrtc.org/2742903002
Cr-Commit-Position: refs/heads/master@{#17326}
2017-03-21 18:04:53 +00:00
zstein
4b2e0829ca Use the same draft version in SDP data channel answers as used in the offer.
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}
2017-02-18 03:48:38 +00:00
deadbeef
a4549d6588 Fix SDP parsing crash due to missing track ID in "a=msid".
BUG=chromium:686405

Review-Url: https://codereview.webrtc.org/2676293003
Cr-Commit-Position: refs/heads/master@{#16545}
2017-02-11 01:26:22 +00:00
deadbeef
90f1e1e0d7 Fixing SDP parsing crash due to invalid port numbers.
BUG=chromium:677029

Review-Url: https://codereview.webrtc.org/2675273003
Cr-Commit-Position: refs/heads/master@{#16541}
2017-02-10 20:35:05 +00:00
ossu
aa4b0775aa Simplify IsFmtpParam according to RFC 4855.
This should help pave the way for injectable audio codecs, since
external implementations need to be able to signal arbitrary fmtp
parameters.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2661453003
Cr-Commit-Position: refs/heads/master@{#16360}
2017-01-30 15:41:18 +00:00
ossu
e1405ad0d1 Removed double-special-casing of ISAC in libjingle and WebRtcVoE.
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.

webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.

Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
2017-01-23 16:55:48 +00:00
ossu
7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.

Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.

Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.

BUG=webrtc:5883

Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 12:56:25 +00:00