22 Commits

Author SHA1 Message Date
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
zhihuang
38ede13042 Support building WebRTC without audio and video.
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}
2017-06-15 19:52:32 +00:00
perkj
77cd58e140 This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.

BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc

Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
2017-05-30 10:52:10 +00:00
perkj
33bb86d5a1 Remove final from RtcEventLogNullImpl
The reason is that there might be implementations that do not want to implement all methods.
To allow easier modification of the RtcEventLog interface, allow these implementation to inherit the RtcEventLogNullImpl implementation.

BUG=none

Review-Url: https://codereview.webrtc.org/2903003002
Cr-Commit-Position: refs/heads/master@{#18298}
2017-05-29 09:46:05 +00:00
perkj
f472699bbd Replace AudioSendStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2856063003
Cr-Commit-Position: refs/heads/master@{#18224}
2017-05-22 17:12:26 +00:00
perkj
ac8f52de70 Replace AudioReceiveStream::Config with rtclog::StreamConfig.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2851303007
Cr-Commit-Position: refs/heads/master@{#18223}
2017-05-22 16:36:28 +00:00
perkj
c0876aab46 Replace VideoSendStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2857933002
Cr-Commit-Position: refs/heads/master@{#18221}
2017-05-22 11:08:28 +00:00
perkj
09e71daec5 Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog.
BUG=webrtc:7538

Review-Url: https://codereview.webrtc.org/2850793002
Cr-Commit-Position: refs/heads/master@{#18220}
2017-05-22 10:26:49 +00:00
michaelt
9765370416 Resolve dependency between rtc_event_log_api and remote_bitrate_estimator
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2800633004
Cr-Commit-Position: refs/heads/master@{#17638}
2017-04-11 07:49:44 +00:00
michaelt
cde46b7278 Resolve cyclic dependency between audio network adaptor and event log api
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2745473003
Cr-Commit-Position: refs/heads/master@{#17565}
2017-04-06 12:59:10 +00:00
philipel
32d0010d86 Add probe logging to RtcEventLog.
In this CL:
 - Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
 - Add corresponding log functions to RtcEventLog.
 - Add optional field |probe_cluster_id| to RtpPacket message and added
   an overload function to log with this information.
 - Propagate the probe_cluster_id to where RTP packets are logged.

BUG=webrtc:6984

Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
2017-02-27 10:18:46 +00:00
terelius
424e6cfd58 Rename some variables and methods in RTC event log.
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).

BUG=webrtc:6423

Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
2017-02-20 13:14:41 +00:00
terelius
0baf55d23b Add logging of delay-based bandwidth estimate.
BUG=webrtc:6423

Review-Url: https://codereview.webrtc.org/2695923004
Cr-Commit-Position: refs/heads/master@{#16663}
2017-02-17 11:38:28 +00:00
minyue
4b7c952376 Reland of "Log audio network adapter decisions in event log."
This was originally reviewed https://codereview.webrtc.org/2559953002/

It was reverted due to a bug in the original CL, see https://codereview.webrtc.org/2631703002/

This CL is to fix and reland.

BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2644863002
Cr-Commit-Position: refs/heads/master@{#16242}
2017-01-24 12:54:59 +00:00
sakal
363a29157a Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
Reason for revert:
Breaks chromium.webrtc.fyi.

Original issue's description:
> Log audio network adapter decisions in event log.
>
> BUG=webrtc:6845
>
> Review-Url: https://codereview.webrtc.org/2559953002
> Cr-Commit-Position: refs/heads/master@{#16053}
> Committed: 3663681b5d

TBR=minyue@webrtc.org,henrik.lundin@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2631703002
Cr-Commit-Position: refs/heads/master@{#16054}
2017-01-13 14:52:12 +00:00
michaelt
3663681b5d Log audio network adapter decisions in event log.
BUG=webrtc:6845

Review-Url: https://codereview.webrtc.org/2559953002
Cr-Commit-Position: refs/heads/master@{#16053}
2017-01-13 14:10:16 +00:00
nisse
306127635e Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.
TBR=pthatcher@webrtc.org
BUG=webrtc:6733

Review-Url: https://codereview.webrtc.org/2515653002
Cr-Commit-Position: refs/heads/master@{#15711}
2016-12-20 13:03:58 +00:00
ossu
f515ab8c3f Moved call.h and most of api/call/* into call/
BUG=webrtc:6716

Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
2016-12-07 12:53:04 +00:00
ivoc
e0928d8002 Added logging for audio send/receive stream configs.
BUG=webrtc:4741,webrtc:6399

Review-Url: https://codereview.webrtc.org/2353543003
Cr-Commit-Position: refs/heads/master@{#14585}
2016-10-10 12:12:57 +00:00
skvlad
cc91d284e4 Moved RtcEventLog files from call/ to logging/
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.

After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).

The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).

This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.

BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2380683005 .

Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-04 01:31:32 +00:00