This layer takes in a simplified "options" struct and the current local description,
and generates a new offer/answer. Previously the options struct assumed there would
only be one media description per media type (audio/video), but it now supports
N number of audio/video descriptions.
The |add_legacy_stream| options is removed from the mediasession.cc/.h
in this CL.
The next step is to add the ability for PeerConnection/WebRtcSession to create
"options" to represent multiple RtpTransceivers, and apply the Unified Plan
descriptions correctly. Right now, only Plan B descriptions will be
generated in unit tests.
BUG=chromium:465349
Review-Url: https://codereview.webrtc.org/2991693002
Cr-Commit-Position: refs/heads/master@{#19343}
Added documentation of thread expectations for video tracks and sources to the API.
Originally landed as patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/.
Patchset 1 is the originall cl.
Patschet 2 is modified so that VideoTrackInterface::AddSink and RemoveSink have a default implementation.
BUG=none
Review-Url: https://codereview.webrtc.org/2989113002
Cr-Commit-Position: refs/heads/master@{#19195}
This allows an application to easily override the default limit
(currently 5).
Also adding a test that covers more of the
PeerConnection<->PortAllocator interaction.
BUG=webrtc:7703
Review-Url: https://codereview.webrtc.org/2985653003
Cr-Commit-Position: refs/heads/master@{#19160}
There is an inconsistency in behavior of PeerConnection.
When I remove track from PeerConnection observer->OnRenegotiationNeeded is called, however if I remove track from MediaStream then there is no notification to renegotiate.
This patch adds missing OnRenegotiationNeeded calls.
BUG=webrtc:7966
Review-Url: https://codereview.webrtc.org/2977493002
Cr-Commit-Position: refs/heads/master@{#19125}
This reverts commit 26d5e2e2809558148dc1e977ec1bc8318a2047bc.
Reverted originally because it dependend on a CL which was reverted. That CL has been reinstated in: https://chromium-review.googlesource.com/#/c/572070/
Bug: webrtc:7969
Change-Id: I404c3a42ad447312d981646dca0aa4cf0ec3134e
Reviewed-on: https://chromium-review.googlesource.com/572403
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19034}
This is a partial re-land. The change doesn't make the default Thread ctor protected anymore but it does mark it as deprecated and updates all use of it in WebRTC.
Original issue's description:
Make the default ctor of rtc::Thread, protected.
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2977953002
Cr-Commit-Position: refs/heads/master@{#19031}
Check that ice_regather_interval_range is set only when continual
regathering is also set.
Bug: webrtc:7969
Change-Id: Ifcfeee744d817cf00914418d7e682f11528faf05
Reviewed-on: https://chromium-review.googlesource.com/569358
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19009}
Reason for revert:
Break projects.
Original issue's description:
> Make the default ctor of rtc::Thread, protected.
> The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
>
> The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2981623002
> Cr-Commit-Position: refs/heads/master@{#19001}
> Committed: a8a3515997TBR=kthelgason@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2979963002
Cr-Commit-Position: refs/heads/master@{#19003}
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2981623002
Cr-Commit-Position: refs/heads/master@{#19001}
Reason for revert:
It breaks a downstream project.
Original issue's description:
> Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
>
> Added documentation of thread expectations for video tracks and sources to the API.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2964863002
> Cr-Commit-Position: refs/heads/master@{#18938}
> Committed: f1377f7222TBR=deadbeef@webrtc.org,noahric@chromium.org,yujo@chromium.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=None
Review-Url: https://codereview.webrtc.org/2979493003
Cr-Commit-Position: refs/heads/master@{#18942}
Added documentation of thread expectations for video tracks and sources to the API.
BUG=None
Review-Url: https://codereview.webrtc.org/2964863002
Cr-Commit-Position: refs/heads/master@{#18938}
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}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
Reason for revert:
Broken downstream project.
Original issue's description:
> Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
>
> BUG=webrtc:7395
>
> Review-Url: https://codereview.webrtc.org/2888303005
> Cr-Commit-Position: refs/heads/master@{#18417}
> Committed: 9641c13327TBR=deadbeef@webrtc.org,stefan@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,holmer@google.com,zstein@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7395
Review-Url: https://codereview.webrtc.org/2914413002
Cr-Commit-Position: refs/heads/master@{#18420}
The only thing the physical socket server was used for was
"Wait"/"WakeUp", but it could be replaced by a simple rtc::Event.
So, removing this dependency makes things less confusing; the fact that
VirtualSocketServer takes a PhysicalSocketServer may lead someone to
think it uses real sockets internally, when it doesn't.
BUG=None
Review-Url: https://codereview.webrtc.org/2883313003
Cr-Commit-Position: refs/heads/master@{#18172}
For every failing test, the solution was either to do a "has IPv4" check
before the test is run, or avoid depending on real network interfaces
altogether.
This specifically fixes rtc_unittests, peerconnection_unittests, and
webrtc_nonparallel_tests.
BUG=None
Review-Url: https://codereview.webrtc.org/2881973002
Cr-Commit-Position: refs/heads/master@{#18155}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
Before this CL, we would negotiate:
- No crypto suites for data m= sections.
- A full set for audio m= sections.
- The full set, minus SRTP_AES128_CM_SHA1_32 for video m= sections.
However, this doesn't make sense with BUNDLE, since any DTLS
association could end up being used for any type of media. If
video is "bundled on" the audio transport (which is typical), it
will actually end up using SRTP_AES128_CM_SHA1_32.
So, this CL moves the responsibility of deciding SRTP crypto suites out
of BaseChannel and into DtlsTransport. The only two possibilities are
now "normal set" or "normal set + GCM", if enabled by the PC factory
options.
This fixes an issue (see linked bug) that was occurring when audio/video
were "bundled onto" the data transport. Since the data transport
wasn't negotiating any SRTP crypto suites, none were available to use
for audio/video, so the application would get black video/no audio.
This CL doesn't affect the SDES SRTP crypto suite negotiation;
it only affects the negotiation in the DLTS handshake, through
the use_srtp extension.
BUG=chromium:711243
Review-Url: https://codereview.webrtc.org/2815513012
Cr-Commit-Position: refs/heads/master@{#17810}
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
Also renames "peerconnection_unittests" to "peerconnection_integrationtests",
and moves the ICE URL parsing code to separate files.
The main problem previously was that the test assertions
occurred in various places in the main test class, and this shared test
code was overly complex and stateful. As a result, it was difficult to
tell what a test even does, let alone what assertions it's meant to be
making. And writing a new test that does what you want can be a
frustrating ordeal.
The new code still uses helper methods, but they have intuitive names
and a smaller role; all of the important parts of the test's logic are
in the test case itself.
We're planning on merging PeerConnection and WebRtcSession at some point
soon, so it seemed valuable to do this, so that the WebRtcSession tests
can be rewritten as PeerConnection tests using better patterns.
BUG=None
Review-Url: https://codereview.webrtc.org/2738353003
Cr-Commit-Position: refs/heads/master@{#17458}
To simplify things, the candidate pool is only used in the first
offer/answer.
After setting a local description, the size is frozen, and changing ICE
servers won't refresh the pool.
After setting an answer, the pooled candidates are discarded.
BUG=webrtc:5180
Review-Url: https://codereview.webrtc.org/2717893003
Cr-Commit-Position: refs/heads/master@{#17178}
This utility class can be used to represent either an error or a
successful return value. Follows the pattern of StatusOr in the protobuf
library.
This will be used by ORTC factory methods; for instance, CreateRtpSender
will either return an RtpSender or an error if the parameters are
invalid or some other failure occurs.
This CL also moves RTCError classes to a separate file, and adds tests
that were missing before.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2692723002
Cr-Commit-Position: refs/heads/master@{#16659}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
Stop the RtcEventLog when the PeerConnection is closed so that Chrome
will not crash because of creating too many threads.
BUG=chromium:687553
Review-Url: https://codereview.webrtc.org/2682433005
Cr-Commit-Position: refs/heads/master@{#16482}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
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}
