79c3359e67
Add end-to-end H.264 packetization test.
...
Also correctly wires up H.264 packetization in the new Call api.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06 09:24:53 +00:00
dde16f19e3
Fix some code styles.
...
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22009004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-05 23:35:43 +00:00
bfe6e08195
Add simulation of network effects to video_loopback tool.
...
Also add support for uniform random packet loss to the fake network pipe.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 12:30:18 +00:00
f9460688a6
Make sure padding is sent on the first sending RTP module.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:41:25 +00:00
f24c4a3b8d
Fix flaky ramp-up test.
...
Don't require the first estimate to be less than the target bitrate. There are other tests verifying that BWE works, so it's enough for this test to measure the
time it takes to ramp-up.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6764 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 10:27:41 +00:00
3c10758b3b
Check before send/receive rtp header extensions.
...
BUG=1788
R=pbos@webrtc.org , tommi@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13949004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-20 15:27:35 +00:00
2f4b14e3f3
Make RTCP sender report send media bytes.
...
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:25:39 +00:00
168f23faa5
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
...
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
bd9c0920ec
Skip encoding in fake VP8 encoder.
...
Broke memcheck, FakeEncoder::Encode doesn't produce valid VP8 frames.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6652 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 13:21:40 +00:00
91f1752f2d
Support VP8 encoder settings in VideoSendStream.
...
Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 10:13:37 +00:00
b8e9e44eac
Add full stack test cases with a fake network pipe.
...
R=pbos@webrtc.org
BUG=1872
Review URL: https://webrtc-codereview.appspot.com/20889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-09 11:29:06 +00:00
3d7da88e06
Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.
...
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 13:59:46 +00:00
62bafae661
Some refactoring inside rtp_rtcp/.
...
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
af38f4e511
Extract RTP-header SSRC inline in Call.
...
Prevents unknown-RTP-header-extension warnings to be flooding from the
RTP-header parsing as there's no way to register RTP extensions for the
parser in Call as they're allowed to differ between RTP streams.
RTP-header parsing should instead be done separately in every
VideoReceiveStream.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 07:38:12 +00:00
161f808500
Add test for VideoEncoder setup/teardown.
...
Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().
BUG=2339
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 14:22:35 +00:00
2bb1bdab8d
Preserve RTP states for restarted VideoSendStreams.
...
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00
b941fe8098
Fix data races related with traces in bitrate estimator test.
...
BUG=3549
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 08:50:48 +00:00
bd249bc711
Remove GetDefaultConfigs() from Call.
...
Defaults for configs are instead placed in the Config constructors.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 04:45:15 +00:00
b9f5453e29
Add boilerplate code for H.264.
...
R=mflodman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 12:42:07 +00:00
20c1f56992
Configure RTX send status on new modules.
...
Fixes bug where newly-allocated modules wouldn't send payload-based
padding (or probably not send over RTX at all).
As the newly-added test exposed lock-inversions shown on tsan in
VideoReceiver, VideoReceiver was thread-annotated and locks taken less.
BUG=chromium:391085
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6601 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 10:58:12 +00:00
614000d638
Adding pbos as video/ owner and removing persons never working with this folder.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6599 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 08:50:14 +00:00
be9d2a4549
Reserve RTP/RTCP modules in SetSSRC.
...
Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-30 13:19:09 +00:00
994d0b7229
Refactor Call-based tests.
...
Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.
BUG=3035
R=kjellander@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-27 08:47:52 +00:00
1227ab89a7
GN: Add BUILD.gn files + kjellander to OWNERS
...
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.
I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.
I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.
BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-23 19:21:07 +00:00
f425b55eeb
Add tests of texture frames in video_send_stream_test.
...
Also fix a bug in ViEFrameProviderBase::DeliverFrame that
a texture frame was only delivered to the first callback.
BUG=chromium:362437
TEST=Run video engine test and webrtc call on CrOS.
R=kjellander@webrtc.org , pbos@webrtc.org , stefan@webrtc.org , wuchengli@google.com
Review URL: https://webrtc-codereview.appspot.com/15789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6506 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 12:04:05 +00:00
eb16b811fb
Implements start bitrate for new video API.
...
Added a new rampup test.
BUG=2879
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 08:57:39 +00:00
cb254aac3b
Enable pacing by default and remove the option to disable it from the new API.
...
BUG=1672
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 15:12:25 +00:00
fbb567dacd
Add APIs to enable padding with redundant payloads.
...
Also makes a small change to the tests to remove flakiness. We can't do
BWE only based on rtp timestamps if we preemptively resend packets instead
of sending padding packets.
BUG=1812,2992
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 13:41:36 +00:00
6ae48c6609
Make VideoSendStream/VideoReceiveStream configs const.
...
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-06 10:49:19 +00:00
eae7924836
Adding back platform specific renderer to video loopback test.
...
BUG=3039
TEST=locally on Mac and Win, video_loopback test
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6339 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 09:32:51 +00:00
ef92755780
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
...
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15629005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:25:29 +00:00
9aa7d8df95
Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky.
...
BUG=3374
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6267 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 05:03:52 +00:00
1457b4737a
First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
...
Receive stats is reset if the payload type changes. Update stats after a possible reset.
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6247 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26 13:06:04 +00:00
1566ee2893
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
...
This reverts commit r6230 to re-land r6229.
ViECapturer::SwapFrame now resets timestamps.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6231 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 13:03:45 +00:00
2cdd433edf
Revert "Remove VideoSendStreamInput::PutFrame."
...
This reverts r6229.
Test WebRtcVideoChannel2BaseTest.MuteStream fails after r6229.
BUG=
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19529005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6230 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 10:43:26 +00:00
f3085e43ab
Remove VideoSendStreamInput::PutFrame.
...
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 09:41:45 +00:00
4059c2f579
Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky.
...
TBR=wu@webrtc.org
BUG=3374
Review URL: https://webrtc-codereview.appspot.com/14579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6201 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:12:29 +00:00
024e4d5c6e
Fix Win VideoSendStream::...::ToString() compiles.
...
Removed an extra ::VideoSendStream in the method declarations.
BUG=3171
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6171 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 10:03:24 +00:00
1e92b0a93d
Add ToString() to VideoSendStream::Config.
...
Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.
BUG=3171
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 09:35:06 +00:00
caba2d2a37
Add DeliveryStatus enum to DeliverPacket().
...
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:57:12 +00:00
db60434b31
Re-enable the BitrateEstimatorTest cases for the Call API.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 09:15:19 +00:00
023b101f4e
Move gflags usage to video_loopback.
...
gflags aren't used by the test environment and is an unnecessary
dependency. They're only used by the video_loopback target, so moving
them there.
R=mflodman@webrtc.org
BUG=3113
Review URL: https://webrtc-codereview.appspot.com/12379006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6120 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 11:26:40 +00:00
66773a032a
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
...
BUG=3111
TEST=try bots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 17:09:44 +00:00
0224c20fa6
* Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter.
...
* Re-enable test CaptureNtpTimeWithNetworkJitter.
* Use 100ms as the threadhold as a FYI since this is a performance test.
BUG=3271
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6054 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-05 17:42:43 +00:00
de1429e9ad
Add thread annotations to Call API.
...
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org , stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 13:00:21 +00:00
c891577e6d
Disable flaky CaptureNtpTimeWithNetworkJitter.
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TBR=wu@webrtc.org
BUG=3271
Review URL: https://webrtc-codereview.appspot.com/19389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5996 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:57:32 +00:00
f043f79711
Disabling flaky CanReceiveFec.
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CanReceiveFec is flaky, likely to the test expecting the first FEC
packet to always be decoded and rendered.
R=stefan@webrtc.org
BUG=3269
Review URL: https://webrtc-codereview.appspot.com/19379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5992 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 09:00:50 +00:00
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
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BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
5ca6a5387e
Remove TraceCallback use from Call.
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Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.
R=mflodman@webrtc.org
BUG=3250,3157
Review URL: https://webrtc-codereview.appspot.com/12329006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 11:35:33 +00:00
