admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
9,643 Commits 1 Branch 0 Tags
Commit Graph

64 Commits

Author SHA1 Message Date
Peter Boström
49e196af40 Remove VideoFrameType aliases for FrameType. 
No longer used in Chromium, so these can now be removed.

BUG=webrtc:5042
R=mflodman@webrtc.org
TBR=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1415693002 .

Cr-Commit-Position: refs/heads/master@{#10390}
 2015-10-23 13:58:27 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h 
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
 2015-10-21 06:00:57 +00:00
asapersson
da535c4055 Add histogram for percentage of sent frames that are limited in resolution due to bandwidth: 
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"

If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"

BUG=

Review URL: https://codereview.webrtc.org/1311533012

Cr-Commit-Position: refs/heads/master@{#10333}
 2015-10-20 06:32:48 +00:00
pbos
22993e1a0c Unify FrameType and VideoFrameType. 
Prevents some heap allocation and frame-type conversion since interfaces
mismatch. Also it's less confusing to have one type for this.

BUG=webrtc:5042
R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1371043003

Cr-Commit-Position: refs/heads/master@{#10320}
 2015-10-19 09:39:15 +00:00
asapersson
4306fc70d7 Add histogram for percentage of sent frames that are limited in resolution due to quality: 
- "WebRTC.Video.QualityLimitedResolutionInPercent"

and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"

BUG=

Review URL: https://codereview.webrtc.org/1325153009

Cr-Commit-Position: refs/heads/master@{#10319}
 2015-10-19 07:35:27 +00:00
Peter Boström
5d0379da2c Remove kSkipFrame usage. 
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.

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

Review URL: https://codereview.webrtc.org/1369923005 .

Cr-Commit-Position: refs/heads/master@{#10181}
 2015-10-06 12:05:03 +00:00
sprang
fb30c1b5d1 Update VP8 settings to avoid spending bitrate on static areas. PERF NOTE 
This CL changes the threshold where we consider a block to be static and
of sufficient quality to not spend bits/CPU encoding it.

Perf note: This change may result in a minor degradation of PSNR/SSIM
and available send bitrate. CPU usage and bitrate sent should however
be greately reduced.

BUG=webrtc:5015

Review URL: https://codereview.webrtc.org/1383533002

Cr-Commit-Position: refs/heads/master@{#10134}
 2015-10-01 13:26:16 +00:00
Peter Boström
1741770742 Implement a high-QP threshold for Android H.264. 
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.

BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1364253002 .

Cr-Commit-Position: refs/heads/master@{#10078}
 2015-09-25 15:03:37 +00:00
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros. 
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
 2015-09-17 07:24:51 +00:00
Asa Persson
110443c1ec Fix for frame resolution in encoded frame callback. 
Scaled resolution for down scaled frames by the quality scaler is not used.

BUG=webrtc:4966
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1317463005 .

Cr-Commit-Position: refs/heads/master@{#9873}
 2015-09-07 13:04:00 +00:00
pbos
ef35f069e7 Remove webrtc::Config from ViEChannelGroup. 
Also removing webrtc/experiments.h which is no longer used.

BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1250513006

Cr-Commit-Position: refs/heads/master@{#9642}
 2015-07-27 15:37:14 +00:00
jackychen
6e2ce6e1ae Allow for framerate reduction for HW encoder. 
R=pbos@webrtc.org, stefan@webrtc.org
TBR=glaznev@google.com

Review URL: https://webrtc-codereview.appspot.com/51159004 .

Cr-Commit-Position: refs/heads/master@{#9573}
 2015-07-13 23:26:40 +00:00
Erik Språng
2c4c914819 In screenshare mode, suppress VP8 bitrate overshoot and increase quality 
This change includes several improvements:

* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers

A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.

BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1193513006.

Cr-Commit-Position: refs/heads/master@{#9495}
 2015-06-24 09:24:50 +00:00
Peter Boström
6a688f5265 Add default downscale threshold to QualityScaler. 
Prevents downscaling below 160x90 or 90x160 to gain more quality.

BUG=4625
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1160403004.

Cr-Commit-Position: refs/heads/master@{#9480}
 2015-06-22 06:03:07 +00:00
Peter Boström
1b9add9df9 Prevent bitrate overshoot for HD layer in VP8. 
BUG=chromium:487648
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55469004

Cr-Commit-Position: refs/heads/master@{#9394}
 2015-06-08 20:52:42 +00:00
Miguel Casas-Sanchez
4765070b8d Rename I420VideoFrame to VideoFrame. 
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.

Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.

BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52629004

Cr-Commit-Position: refs/heads/master@{#9339}
 2015-05-30 00:21:56 +00:00
Stefan Holmer
e87d48719f Fix ARM64 detection for VP8 and VP9 wrappers. 
BUG=4702
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57459004

Cr-Commit-Position: refs/heads/master@{#9287}
 2015-05-26 20:10:38 +00:00
jackychen
98d8cf58ee Hardware VP8 encoding: Use QP as metric for resize. 
Add vp8 frame header parser to get QP from vp8 bitstream.

BUG= 4273
R=glaznev@webrtc.org, marpan@google.com, pbos@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49259004

Cr-Commit-Position: refs/heads/master@{#9256}
 2015-05-21 18:11:53 +00:00
Peter Boström
b302ad4eab Remove unused VideoDecoder methods. 
Removing VideoDecoder::Copy() and
VideoDecoder::SetCodecConfigParameters().

Also adding override to VP8DecoderImpl.

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

Review URL: https://webrtc-codereview.appspot.com/55409004

Cr-Commit-Position: refs/heads/master@{#9244}
 2015-05-21 07:42:14 +00:00
Erik Språng
143cec1cc6 Set correct encoder-specific settings for vpx in the new API. 
Also, make VideoEncoderConfig::ContentType an enum class.

BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46069004

Cr-Commit-Position: refs/heads/master@{#9093}
 2015-04-28 08:01:14 +00:00
jackychen
61b4d518af Dynamic resolution change for VP8 HW encode. 
Off by default for now.

BUG=
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45849004

Cr-Commit-Position: refs/heads/master@{#9045}
 2015-04-21 22:29:53 +00:00
pbos@webrtc.org
a3209a2b27 Release buffer pool in Vp8DecoderImpl::Release(). 
Permits reusing an external VP8DecoderImpl instance from another
VideoReceiveStream without a thread-checker DCHECK blowing up. Also
releases buffers that would've been kept in memory even though the
decoder isn't configured.

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

Review URL: https://webrtc-codereview.appspot.com/50449004

Cr-Commit-Position: refs/heads/master@{#8807}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 13:36:25 +00:00
magjed@webrtc.org
e155dbeae9 VP8/9EncoderImpl::Encode: Check resolution of input I420VideoFrame 
This CL adds checks in Encode to guard against memory reads out of bounds.

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

Review URL: https://webrtc-codereview.appspot.com/46429008

Cr-Commit-Position: refs/heads/master@{#8750}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 12:27:40 +00:00
magjed@webrtc.org
73d763e71f Add I420 buffer pool to avoid unnecessary allocations 
Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well.

BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41189004

Cr-Commit-Position: refs/heads/master@{#8748}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 11:41:15 +00:00
magjed@webrtc.org
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." 
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:48:18 +00:00
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
marpan@webrtc.org
6daacbc8ae Set cpu_speed parameter for low resolutions, for non-simulcast. 
Allow for setting different cpu_speed setting based on resolution, for non-simulcast.
Use the existing low resolution simulcast cpu_speed setting for the non-simulcast case.

No change to simulcast behavior, unless top/highest layer stream is also below CIF resolution,
(in which case all layers will use lower the cpu_speed setting =-4).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37319004

Cr-Commit-Position: refs/heads/master@{#8603}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 21:47:18 +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
pbos@webrtc.org
9115cde6c9 Merge VP8 changes. 
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:36:40 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
johannkoenig@google.com
94ff92ceec Use VPX_IMG_FMT_*/VPX_PLANE_* defines 
The compatibility layer has been removed upstream:
https://gerrit.chromium.org/gerrit/gitweb?p=webm%2Flibvpx.git;a=commit;h=9cdaa3d72eade9ad162ef8f78a93bd8f85c6de10

BUG=webrtc:3839
R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 17:31:47 +00:00
pbos@webrtc.org
a0d7827b16 Add ability to downscale content to improve quality. 
BUG=3712
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 11:51:47 +00:00
niklas.enbom@webrtc.org
9c09e6ee2b Add high perf mode to VP8 
R=marpan@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 16:32:08 +00:00
wu@webrtc.org
6c75c98964 Propagate capture ntp timestamp from rtp to renderer. 
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.

TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 17:46:33 +00:00
wu@webrtc.org
1faef7d084 Use codec width/height as the encoded_image width/height. 
The raw_->w and raw_->h which are the stored image width/height may not be the encoded image size in the case when the incoming frame has a odd size.

R=marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-20 17:00:46 +00:00
wu@webrtc.org
7f52a6ef2b Split the implementation of VP8Encoder|Decoder::Create into a seperated file 
(vp8_factory.cc).

R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 23:56:39 +00:00
andresp@webrtc.org
5500d93fe5 Add temporal layer factory. 
R=marpan@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2180004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-06 11:26:15 +00:00
mikhal@webrtc.org
b2c28c3699 Relanding 4597 - Don't force key frame when decoding with errors. 
Makes sure that incomplete key frame or delta frames will be released from the JB when decoding with errors.
The decoder in turn will trigger a PLI until a complete key frame is received in order to start a session.

TBR=stefan@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/2097004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-23 21:54:50 +00:00
henrike@webrtc.org
ceea41d135 Revert 4597 "Don't force key frame when decoding with errors" 
> Don't force key frame when decoding with errors
> 
> BUG=2241
> R=stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2036004

TBR=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2093004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-23 00:53:24 +00:00
mikhal@webrtc.org
44af55cc44 Don't force key frame when decoding with errors 
BUG=2241
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2036004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-22 23:29:43 +00:00
tnakamura@webrtc.org
d2102afa2a Undo libvpx include changes in r4348 to fix build. 
A longer term fix is needed, but this at least quickly unblocks the build.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1816005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 18:48:24 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:32:05 +00:00
fbarchard@google.com
d7148c86c5 Use 3 threads for higher than 720p resolutions 
BUG=1893
TEST=untested
R=ajm@google.com, andrew@webrtc.org, dingkai@google.com, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1684004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-18 22:06:42 +00:00
hclam@chromium.org
b3e5acfb66 Cleanup traces in WebRTC 
Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:13:02 +00:00
fbarchard@google.com
1e3c794688 Use 2 threads for HD, or 1 for VGA or less. 
BUG=1739
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1438005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 18:43:38 +00:00
pbos@webrtc.org
d25b602dc0 VP8: Avoid copying the codec struct on Reset(). 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1319013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-23 13:08:04 +00:00
andresp@webrtc.org
48c5882f2a Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. 
R=stefan@webrtc.org,marpan@webrtc.org
BUG=201

Review URL: https://webrtc-codereview.appspot.com/1323005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 15:31:40 +00:00
marpan@webrtc.org
557e92515d Reapply the reverted r3747. 
https://code.google.com/p/webrtc/source/detail?r=3747

r3747 timed-out on a tsan test. Verified that it passes
the test and reduced the execution time of that test (r3782).

TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1292006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 21:21:32 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00