Commit Graph

106 Commits

Author SHA1 Message Date
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
stefan@webrtc.org
0b38478885 Add support for parsing header only RTP dumps with bwe_rtp_play.
Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:43:49 +00:00
pbos@webrtc.org
9f79fe684a Merge remote bitrate estimator changes.
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:34:06 +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
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
stefan@webrtc.org
4abadab708 Simplify bwe tests.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7576 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 10:47:12 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms.
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 12:57:56 +00:00
stefan@webrtc.org
82462aade0 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
Also wires up a finch experiment to control this.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +00:00
kjellander@webrtc.org
b8caf6a504 GN: Enable libvpx, add link target and convert some test targets
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).

I also converted a few test targets and made a GN file for
third_party/gflags.

BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 18:05:02 +00:00
kjellander@webrtc.org
f21ea918ad GN: Add common configs to all targets.
This is needed to ensure we have the same build with GN
as with GYP, since GYP includes the common.gypi on a global level.
Several fixes has been needed in the past because some code have
been built without the right defines.

BUG=3441
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7317 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 17:37:22 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths.
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 09:55:40 +00:00
kjellander@webrtc.org
b96ea2aab5 Remove former team members from OWNERS and WATCHLISTS
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 06:12:08 +00:00
kjellander@webrtc.org
42ee5b54b5 GN: Disable Chromium clang plugins for standalone build.
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.

The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.

BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 14:15:35 +00:00
pbos@webrtc.org
4b5625e5ac RTP video playback tool using Call APIs.
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06 16:26:56 +00:00
kjellander@webrtc.org
9bef551ba1 GN: Fix include paths for WebRTC in Chromium build.
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.

This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.

However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.

BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-13 09:02:54 +00:00
stefan@webrtc.org
bee164a214 Fix test issues and a win compile error introduced with r6605.
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
  paced_sender.h
    tick_util.h
      windows.h
        windef.h

And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:00:06 +00:00
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 09:20:42 +00:00
stefan@webrtc.org
9138eb649b Fix compile error introduced with r6571.
TBR=mflodman@webrtc.org

BUG=3527

Review URL: https://webrtc-codereview.appspot.com/20799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6572 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:44:05 +00:00
stefan@webrtc.org
5779ca478d Fixes a potential BWE clock mismatch bug.
Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.

This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.

BUG=3527
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 12:14:49 +00:00
kjellander@webrtc.org
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
henrike@webrtc.org
88fbb2d86b Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
Same as https://webrtc-codereview.appspot.com/19519004. The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing...
(tested locally).

BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
mcasas@webrtc.org
2fa7f79094 Revert 6202 "Switch to using base/constructormagic.h and remove ..."
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
> 
> BUG=N/A
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19519004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
henrike@webrtc.org
125ffd709d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
andresp@webrtc.org
a36ad6929d Add webrtc field trials API.
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
solenberg@webrtc.org
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
wu@webrtc.org
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
henrike@webrtc.org
f2aafe4355 Added include of assert.h for files calling assert but missing the include.
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
pbos@webrtc.org
0300939484 Disable failing GoogleWifiTrace3Mbps.
Disables BweFeedbackTest.GoogleWifiTrace3Mbps instead of
BweSimulation.GoogleWifiTrace3Mbps.

TBR=stefan@webrtc.org
BUG=3277

Review URL: https://webrtc-codereview.appspot.com/20389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 15:25:59 +00:00
pbos@webrtc.org
9353e6bc55 Disable GoogleWifiTrace3Mbps.
Breaks bots, according to stefan@ there's a missing file for this test
to run.

TBR=stefan@webrtc.org
BUG=3277

Review URL: https://webrtc-codereview.appspot.com/13409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:49:56 +00:00
stefan@webrtc.org
dfe2a1c995 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6015 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:21:42 +00:00
andrew@webrtc.org
8f69330310 Replace scoped_array<T> with scoped_ptr<T[]>.
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar...

except for the few not-built-on-Linux files which were updated manually.

TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
stefan@webrtc.org
0175d76c72 Fix leak in remote bitrate estimator tests introduced in r5980
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 11:38:57 +00:00
stefan@webrtc.org
4f616a02bd Support for simulating multiple independent flows in a network.
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 10:59:24 +00:00
wu@webrtc.org
cd70119a10 Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
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
kjellander@webrtc.org
7de47bce12 Remove use of tmpnam.
This solves compilation with the Mac SDK 10.9.

BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org, henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
stefan@webrtc.org
b08db28958 Clean up traces and logs in RemoteBitrateEstimator.
BUG=3153
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5854 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-07 12:53:28 +00:00
stefan@webrtc.org
f8f7c8b618 Added simulations of capacity variations and wifi recordings.
Also changes the packet sizes for the video sender and the trace based filter to match.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5824 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-01 14:00:05 +00:00
solenberg@webrtc.org
caeae4680c Add tests for the RBE RemoveStream() API.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-31 13:33:39 +00:00
stefan@webrtc.org
7e3ee8362b Disabled some of the remote bitrate estimator baseline tests.
These are disabled temporarily until updated.

R=solenberg@webrtc.org
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5758 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:45:13 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:38:25 +00:00
stefan@webrtc.org
af839b28b0 Add AIMD option to BWE API.
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10319005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 09:42:08 +00:00
stefan@webrtc.org
41689018a0 Add max delay to trace based filters and enhances drop tail queues with delay statistics.
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5696 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-13 15:01:32 +00:00
stefan@webrtc.org
9b5f4d8a84 Fix build breakage introduce with r5665.
TBR=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5666 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 09:38:39 +00:00
stefan@webrtc.org
f9e7c9d865 Add option to bwe_rtp_to_text to output arrival times only in nanoseconds.
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5665 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10 09:11:21 +00:00
pbos@webrtc.org
3ecc162d01 Remove std:: prefixes from C functions in webrtc/.
std::memcpy -> memcpy for instance. This change was motivated by a
compile report complaining that std::rand() was used instead of rand(),
probably with a stdlib.h include instead of cstdlib. Use of C functions
without the std:: prefix is a lot more common, so removing std:: to
address this.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5658 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07 15:23:34 +00:00
stefan@webrtc.org
45846977f9 Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted.
R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 15:46:46 +00:00
pbos@webrtc.org
0117d1c48c Fix compilation errors under clang 3.5.
Enables building tip-of-tree clang which introduces new warnings that
cause compilation errors in our code base (-Werror).

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

Review URL: https://webrtc-codereview.appspot.com/9319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-03 16:47:03 +00:00
jiayl@webrtc.org
de782180b0 Change the type of propagation delta from int64 to int.
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.

BUG=2910
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-12 19:19:23 +00:00
jiayl@webrtc.org
1f64f06784 Add stats of incoming frame delays for debugging bandwidth estimation.
BUG=crbug/338380
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10 19:12:14 +00:00