zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
14,320 Commits 1 Branch 0 Tags
43cb716e55eab1cf1ae8afd2ac79b51a604d0fa5
Commit Graph

895 Commits

Author SHA1 Message Date
hbos@webrtc.org
93d9d6503e I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45629004

Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:26:41 +00:00
hbos@webrtc.org
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:02:19 +00:00
pbos@webrtc.org
86639737b8 Remove thread id from ThreadWrapper::Start(). 
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.

BUG=4413
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43699004

Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 00:07:45 +00:00
guoweis@webrtc.org
fdd1057949 Add CVO support to Vie layer. 
1. standard plumbing CVO through vie layer.
2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation.

WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420.

BUG=4145
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429007

Cr-Commit-Position: refs/heads/master@{#8703}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:51:50 +00:00
magjed@webrtc.org
0d9bb8e499 Remove the need for scoped_ptr<I420VideoFrame> in VieCapturer. 
Remove the need for scoped_ptr<I420VideoFrame> in VieCapturer.
This adds the method I420VideoFrame::Reset and replace the use of scoped_ptr in ViECapturer.
Also, a unittest is added to check that ViECapturer does not retain a frame after it has been delivered.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43669004

Cr-Commit-Position: refs/heads/master@{#8678}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 10:07:15 +00:00
magjed@webrtc.org
d7452a0168 Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame." 
This reverts commit r8633.

Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.

BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj

Review URL: https://webrtc-codereview.appspot.com/46549004

Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 15:13:13 +00:00
perkj@webrtc.org
bcead305a2 Make the entry point for VideoFrames to webrtc const ref I420VideoFrame. 
This removes the none const pointer entry and SwapFrame.

Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.

With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame

BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46429004

Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 12:38:22 +00:00
magjed@webrtc.org
45cdcce5f5 Remove TextureVideoFrame 
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.

R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/40229004

Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 10:41:47 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
pbos@webrtc.org
982cd2a94c Filter receiver-side DataCountersUpdated on SSRC. 
BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44509004

Cr-Commit-Position: refs/heads/master@{#8575}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-03 15:57:18 +00:00
stefan@webrtc.org
48ac226b9a Add support for writing h264 decoder input to file and parsing interleaved length/packet RTP dumps. 
This is useful for debugging h264 input when we don't have an h264 decoder, as the resulting file should be possible to play back using mplayer. It is also often convenient to dump rtp packets in an interleaved format where the size of a packet is inserted before the actual payload.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42139004

Cr-Commit-Position: refs/heads/master@{#8558}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-02 16:19:15 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
kwiberg@webrtc.org
ac2d27d9ae Fix style violations in common_types.h and config.h 
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.

The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.

BUG=163
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26089004

Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:01:28 +00:00
pbos@webrtc.org
891d48393e Wire up target_media_bitrate in VideoSendStream. 
Also wires up target_enc_bitrate in WebRtcVideoEngine2.

BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42479004

Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 13:16:17 +00:00
pbos@webrtc.org
3e6e271ec3 Implement CpuOveruseMetrics as callbacks. 
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.

BUG=1667, 1788
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42429004

Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 12:20:24 +00:00
asapersson@webrtc.org
5c928ebd1d Let first packet through to avoid getting key frame requests (and no nacks) for EndToEndTest.ReceivedFecPacketsNotNacked. 
BUG=4328
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38259004

Cr-Commit-Position: refs/heads/master@{#8502}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-25 11:47:43 +00:00
pbos@webrtc.org
09c77b95bb Add decoder-timing stats to VideoReceiveStream. 
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788

Review URL: https://webrtc-codereview.appspot.com/40819004

Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-25 10:42:45 +00:00
pbos@webrtc.org
49096de442 DCHECK send DataCountersUpdated for valid SSRCs. 
Also updates RTPSender to not update RTX stats when RTX is disabled.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42399004

Cr-Commit-Position: refs/heads/master@{#8489}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-24 22:38:22 +00:00
pbos@webrtc.org
8278c072b6 Enable NACK under SendsAndReceivesH264. 
Decoding with errors has a bug that triggers an assert during packet
loss. Switching to NACK since that is what we expected to be running.

BUG=4337
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43399004

Cr-Commit-Position: refs/heads/master@{#8458}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 11:12:18 +00:00
sprang@webrtc.org
343096ac03 Fix incorrect rtx config in full_stack tests. 
BUG=4326
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40819006

Cr-Commit-Position: refs/heads/master@{#8455}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 08:34:48 +00:00
asapersson@webrtc.org
1467421646 Fix for flaky test: VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent. 
Only compare media bytes sent if number of sent packets in rtcp packet are equal to sent rtp packets.

BUG=4327
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34299004

Cr-Commit-Position: refs/heads/master@{#8454}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 08:14:46 +00:00
sprang@webrtc.org
dc77d7447e Disable FullStackTest.ForemanCifPlr5 temporarily while investigating flakiness. 
BUG=4326
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37359004

Cr-Commit-Position: refs/heads/master@{#8442}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-20 10:40:41 +00:00
pbos@webrtc.org
1d0fa5d352 Add RtcpPacketTypeCounter stats to new API. 
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/37489004

Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-19 12:47:45 +00:00
sprang@webrtc.org
131bea89d6 Offline screenshare quality test, plus loopback. 
BUG=4171
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34109004

Cr-Commit-Position: refs/heads/master@{#8408}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 12:46:44 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset(). 
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:58:38 +00:00
pbos@webrtc.org
40367f984b Remove default video encoders for new video API. 
Reduces stream creation time significantly. As a side effect also
removes default encoders for receive-only channels.

BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37049004

Cr-Commit-Position: refs/heads/master@{#8356}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 08:00:42 +00:00
pbos@webrtc.org
9e4e524f38 Use an external-only VideoRenderModule in Call. 
The default render module instantiated from inside VideoEngine if none
exists instantiates platform-specific code. Call only uses external
rendering, so this is an unneccessary overhead.

BUG=1667
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39069004

Cr-Commit-Position: refs/heads/master@{#8346}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-12 10:48:55 +00:00
pbos@webrtc.org
fbcb5ceb16 Remove VideoSendStreamTest.ProducesStats. 
This test is covered by EndToEndTests.GetStats and there's no need for a
duplicate test.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39049004

Cr-Commit-Position: refs/heads/master@{#8332}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-11 14:25:23 +00:00
pbos@webrtc.org
f4c10d24dc Always use DeliverI420Frame in WebRtcVideoEngine. 
Moves native_handle() path to DeliverI420Frame and CHECKs that
DeliverFrame is not being used anymore.

R=magjed@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/38019004

Cr-Commit-Position: refs/heads/master@{#8312}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-10 10:20:38 +00:00
pbos@webrtc.org
0d852d5c27 Use VideoReceiveStream as an ExternalRenderer. 
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.

Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.

R=stefan@webrtc.org
TBR=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/34169004

Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-09 15:15:24 +00:00
pbos@webrtc.org
50fe359eb6 Add tracing for slow paths in new video API. 
Allows tracking what actually takes time in SetRemoteDescription and
SetLocalDescription.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38809004

Cr-Commit-Position: refs/heads/master@{#8202}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-29 12:33:42 +00:00
asapersson@webrtc.org
37c0559c1e Notify jitter buffer about received FEC packets (to avoid sending NACK request for these packets). 
Don't copy codec specific header for empty packets in the jitter buffer.

BUG=3135
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37659004

Cr-Commit-Position: refs/heads/master@{#8184}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:58:40 +00:00
sprang@webrtc.org
2a6558c2a5 Make sure ByteReader<T>::Read* is properly constified. 
Also, start using it in real code...

BUG=
R=holmer@google.com, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37809004

Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 12:38:16 +00:00
asapersson@webrtc.org
cfd82dfc11 Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. 
Prepares for adding FEC bytes to the StreamDataCounter.

R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 09:39:59 +00:00
pbos@webrtc.org
32e8528581 Log configs when creating video streams in Call. 
Adds VideoReceiveStream::Config::ToString and logs configs in both
Call::CreateVideoSendStream and Call::CreateVideoReceiverStream.

R=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/41519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 10:09:39 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
stefan@webrtc.org
8817256373 Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator. 
BUG=chromium:444023
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 18:00:21 +00:00
pbos@webrtc.org
5570769210 Remove the last getters from VideoReceiveStream stats. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/32899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:45:03 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API. 
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:33:17 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
pbos@webrtc.org
98c04b38a8 Get avg_delay_ms from DecoderTiming callback. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:12:52 +00:00
pbos@webrtc.org
2b19f06312 Wire up RTT statistics to webrtc::Call. 
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 13:26:09 +00:00
pbos@webrtc.org
85bd53e7c9 Add AbsSendTime unittests to rampup_tests.cc. 
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 10:36:20 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream. 
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:23:21 +00:00
asapersson@webrtc.org
d952c40c7e Add receive bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 07:38:56 +00:00
henrik.lundin@webrtc.org
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader 
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 15:50:30 +00:00
pbos@webrtc.org
008731868a Implement settable min/start/max bitrates in Call. 
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 14:03:34 +00:00
pbos@webrtc.org
b951eb12c9 Add back EXPECT_TRUEs. 
These shouldn't fail, but EXPECT_TRUE gives nicer error messages that
work in Release. These changes got through unreviewed in r7726.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 11:13:28 +00:00