decurtis@webrtc.org
e2506670a4
Add RefCounting for TransportProxies
...
BUG=1574
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37869004
Cr-Commit-Position: refs/heads/master@{#8238}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8238 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 23:19:23 +00:00
pthatcher@webrtc.org
af01d93aa2
Cleanup and prepare for bundling.
...
- Add a GetOptions function. Needed for eventual bundle testing to
confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.
BUG=1574
R=pthatcher@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39699004
Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 23:14:18 +00:00
decurtis@webrtc.org
322a564f49
Fix datachannel stats id and timestamp.
...
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel.
Adds a timestamp to the data channel reports.
Implements unit tests to verify that the timestamp is set correctly.
BUG=1805
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33119004
Cr-Commit-Position: refs/heads/master@{#8236}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 22:10:13 +00:00
pkasting@chromium.org
0e81fdf5d2
Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting.
...
BUG=chromium:82439
TEST=none
R=henrik.lundin@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40569004
Cr-Commit-Position: refs/heads/master@{#8229}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8229 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 23:54:40 +00:00
pkasting@chromium.org
19f3f71c98
Fix apparent typo: int -> char.
...
The surrounding similar methods all used unsigned char, using unsigned int in
this case looks like an accident, especially since the function passes on the
value in question to a function expecting a uint8.
BUG=none
TEST=none
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40529004
Cr-Commit-Position: refs/heads/master@{#8228}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-02 19:44:42 +00:00
pkasting@chromium.org
026b892e72
Using << on an int8_t or uint8_t will output a character rather than a number.
...
Places that do this need to cast to int to get the desired behavior.
BUG=none
TEST=none
R=henrik.lundin@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40579004
Cr-Commit-Position: refs/heads/master@{#8223}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:54:19 +00:00
pkasting@chromium.org
005b6fffe6
Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
...
BUG=none
TEST=none
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39649004
Cr-Commit-Position: refs/heads/master@{#8222}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8222 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:42:17 +00:00
pbos@webrtc.org
5e161616b1
Remove CPU monitor from WebRtcVideoEngine2.
...
CPU adaptation is based on timings done inside webrtc, not actual CPU
values anymore. This code has never been wired up and is causing flakes
on at least valgrind, but possibly also on actual platforms.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34089004
Cr-Commit-Position: refs/heads/master@{#8221}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8221 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 15:31:26 +00:00
tommi@webrtc.org
aef0779dab
Rewrite ThreadWindows.
...
* Remove "dead" and "alive" variables.
* Remove critical section
* Skip synchronizing with the worker thread to verify startup (no need).
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
Also added some TODOs for myself for the ThreadWrapper interface.
I'm removing the HasNoMonitorThread test since it is no longer relevant and ends up checking the wrong thing (ProcessThread - a generic thread type) in the wrong way (parsing a debug log) :) I think it served a purpose some years ago, but things have changed since.
BUG=2902
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37069004
Cr-Commit-Position: refs/heads/master@{#8220}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8220 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 15:06:44 +00:00
braveyao@webrtc.org
8820ac7cc4
peerconnectin_server: missing comma in sprintfn() in r8128
...
BUG=4244
TEST=Manual Test
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37079004
Cr-Commit-Position: refs/heads/master@{#8213}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8213 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 09:58:45 +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
tommi@webrtc.org
4161715e3f
Remove ChangeUniqueID.
...
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.
It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.
BUG=
R=henrika@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37849004
Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
magjed@webrtc.org
a26f511dd2
Remove frame copy in ViEExternalRendererImpl::RenderFrame
...
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128,4227
R=mflodman@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8136
Review URL: https://webrtc-codereview.appspot.com/36489004
Cr-Commit-Position: refs/heads/master@{#8199}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8199 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 11:45:43 +00:00
braveyao@webrtc.org
a742cb1f37
Enable DTLS for peerconnection example. If it's a loopback test, then we recreate another peerconnection with DTLS off.
...
BUG=3872
TEST=Manual Test
R=jiayl@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36989004
Cr-Commit-Position: refs/heads/master@{#8193}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8193 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 04:23:39 +00:00
pkasting@chromium.org
e7a4a12f83
Add arraysize() macro from Chromium, and make use of it in a few places.
...
This not only shortens some test code, it makes it more robust against changing
the lengths of the arrays later and forgetting to update the length constants
(which bit me).
BUG=none
TEST=none
R=hta@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34829004
Cr-Commit-Position: refs/heads/master@{#8191}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8191 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 21:37:13 +00:00
honghaiz@google.com
a67ca1a3bb
Only report the first rtp packet because it indicates the media has started flowing.
...
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37829004
Cr-Commit-Position: refs/heads/master@{#8189}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8189 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 19:48:40 +00:00
tkchin@webrtc.org
36401aba62
Update GAE API paths for join/leave.
...
BUG=4221
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33069004
Cr-Commit-Position: refs/heads/master@{#8174}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8174 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 21:35:16 +00:00
magjed@webrtc.org
fc5ad95fec
Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139
...
Link to original CL: https://review.webrtc.org/36909004/
R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227
Review URL: https://webrtc-codereview.appspot.com/39669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 09:57:01 +00:00
glaznev@webrtc.org
8501ee632b
Support VP8 HW decoding on devices with Exynos codec.
...
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 23:07:19 +00:00
glaznev@webrtc.org
82415e395f
Update AppRTCDemo to use renamed GAE messages.
...
BUG=4221
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 22:22:50 +00:00
tkchin@webrtc.org
7519de519e
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
...
> Remove frame copy in ViEExternalRendererImpl::RenderFrame
>
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
>
> BUG=1128
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36489004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 21:20:41 +00:00
tkchin@webrtc.org
0f98844749
Revert 8139 "Implement elapsed time and capture start NTP time e..."
...
> Implement elapsed time and capture start NTP time estimation.
>
> These two elements are required for end-to-end delay estimation.
>
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36909004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 21:17:38 +00:00
jiayl@webrtc.org
dacdd9403d
Reland r7980:
...
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908
BUG=4068, crbug/446908
R=juberti@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 17:33:34 +00:00
pbos@webrtc.org
ad3ee2c46b
Implement elapsed time and capture start NTP time estimation.
...
These two elements are required for end-to-end delay estimation.
BUG=1788
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8139 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 14:55:00 +00:00
kjellander@webrtc.org
a02d76845f
Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
...
Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.
Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.
BUG=4219
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 14:34:52 +00:00
magjed@webrtc.org
182ea46fac
Remove frame copy in ViEExternalRendererImpl::RenderFrame
...
Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
BUG=1128
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8136 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 11:50:13 +00:00
tommi@webrtc.org
586f2eda0d
Change GetStreamBySsrc to not copy StreamParams.
...
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple. Also, we can use lambdas now :)
BUG=
R=perkj@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 23:00:41 +00:00
jlmiller@webrtc.org
b40c7bb53c
Change sprintf use in talk samples to snprintf
...
BUG=2301
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8128 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 18:49:06 +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
jiayl@webrtc.org
cceb166a3f
Fix a use-after-free when sending queued messages is aborted for blocked channel.
...
BUG=4187
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 00:55:10 +00:00
tommi@webrtc.org
4fb7e25843
Update StatsReport and by extension StatsCollector to reduce data copying.
...
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.
BUG=2822
R=hta@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 11:36:18 +00:00
braveyao@webrtc.org
fedb9ea6bc
Correct the class name in peerconnection_jni.cc.
...
BUG=4194
TEST=Manual Test
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8106 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 07:57:06 +00:00
jlmiller@webrtc.org
5f93d0a140
Update libjingle license statements at top of talk files for consistency
...
BUG=2133
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
kjellander@webrtc.org
853049fa30
Move internal capture+render to build_with_chromium==0 condition
...
This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).
Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/
TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 11:40:45 +00:00
tommi@webrtc.org
8e327c45d0
Update StatsCollector's interface in preparation of more changes.
...
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=8095
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 20:41:26 +00:00
tommi@webrtc.org
43e54e36bf
Revert 8095 "Update StatsCollector's interface in preparation of..."
...
> Update StatsCollector's interface in preparation of more changes.
>
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
>
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
>
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
>
> BUG=2822
> R=perkj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36829004
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 17:34:23 +00:00
tommi@webrtc.org
5b76fd79df
Update StatsCollector's interface in preparation of more changes.
...
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
BUG=2822
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 16:49:33 +00:00
phoglund@webrtc.org
f9d3555ec3
Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
...
The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.
BUG=None
TESTED=locally
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 13:57:59 +00:00
sprang@webrtc.org
ff9462eb54
Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.
...
Tests are flaky on tsan, disabling for now.
BUG=4135
R=kjellander@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-19 12:06:35 +00:00
decurtis@webrtc.org
487a444215
Add stats collection for the data channel.
...
BUG=1805
R=bemasc@chromium.org , hta@webrtc.org , pthatcher@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 22:55:07 +00:00
tkchin@webrtc.org
ef2a5dd398
Update AppRTCDemo UI.
...
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 22:38:21 +00:00
guoweis@webrtc.org
61c1247224
Fix a case where empty candidate id is used
...
BUG=4161
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-15 06:53:07 +00:00
pthatcher@webrtc.org
fd630a50d2
Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
...
R=decurtis@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 23:19:06 +00:00
pbos@webrtc.org
f1c8b90520
Remove WebRtcVideoEncoderFactory2.
...
This interface is no longer required and just adds complexity.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/33009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 17:29:27 +00:00
pbos@webrtc.org
f18fba2f7b
Implement SimulcastEncoderAdapter support.
...
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/37589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:26:23 +00:00
henrik.lundin@webrtc.org
8315d7de85
Remove dual stream functionality in VoiceEngine
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.
BUG=3520
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:07:26 +00:00
mflodman@webrtc.org
b4e5d1b34e
Remove RTX SSRC when deleting the default receive stream.
...
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 15:07:07 +00:00
kwiberg@webrtc.org
2ebfac5649
Remove COMPILE_ASSERT and use static_assert everywhere
...
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.
R=aluebs@webrtc.org , andrew@webrtc.org , hellner@chromium.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 10:51:54 +00:00
andresp@webrtc.org
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
phoglund@webrtc.org
ef090927f4
No longer asserting in mocks, split first test case in two methods.
...
This way assertions will be caught in the test runner instead of crashing other Android threads.
BUG=None
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8054 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 08:56:06 +00:00
kwiberg@webrtc.org
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +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
glaznev@webrtc.org
be40eb0579
Allow 720x1280 frames encoding on Android.
...
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:55:47 +00:00
perkj@webrtc.org
81134d019d
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.
...
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory.
This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.
Note that both Chrome and the Android implementation use an external signaling thread.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +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
glaznev@webrtc.org
80452d70cb
Sync Android AppRTCDemo with internal repo.
...
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.
R=jiayl@webrtc.org , wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:34:06 +00:00
pthatcher@webrtc.org
9657265f39
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
...
This reverts r7980.
It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.
Review URL: https://webrtc-codereview.appspot.com/41429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:08:27 +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
decurtis@webrtc.org
2ead571fb6
Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
...
BUG=3996
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 19:18:01 +00:00
pbos@webrtc.org
59062d5aef
Rename SendAndReceiveH264SvcQqvga to VP8 instead.
...
This looks like it's been incorrect for a while, this test configures
VP8 in QQVGA.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8018 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 19:21:18 +00:00
decurtis@webrtc.org
8af11042cb
Avoid reading past end of string in GetLine.
...
BUG=3881
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 19:15:51 +00:00
pbos@webrtc.org
bab79951ca
Convert FileMediaEngineTest to use more expects.
...
Allows pinpointing more precisely where a failure occurs.
BUG=4144
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8015 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 18:01:29 +00:00
kjellander@webrtc.org
07c83a1385
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2)
...
In https://webrtc-codereview.appspot.com/35669004/ the wrong
define was used (OS_WIN only exists in Chromium code).
BUG=4135
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8008 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 10:36:53 +00:00
tkchin@webrtc.org
4e5115ae73
RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
...
There should be no change in behavior, since this is what the default
constructor does.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 06:35:18 +00:00
glaznev@webrtc.org
f6a9714760
Remove peer connection and signaling calls from UI thread.
...
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.
R=jiayl@webrtc.org , wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 22:24:09 +00:00
kjellander@webrtc.org
d95435c17a
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win
...
These tests have turned out to be flaky on Windows.
BUG=4135
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8004 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 11:01:35 +00:00
kjellander@webrtc.org
cbe7ca8796
Roll chromium_revision 8e72e1d..271c6cc (307131:309333)
...
This enables OpenSSL by default for Windows, see
8e72e1d..271c6cchttps://codereview.chromium.org/694643002 .
New Android dependencies are required due to being introduced in
build/android/pylib/remote/device/remote_device_test_run.py
of 5c49978f09https://codereview.chromium.org/815213002 was submitted, since
it's based on e2a338fac98e72e1d..271c6cc8e72e1d..271c6ccR=pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8003 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 07:24:27 +00:00
tkchin@webrtc.org
3a63a3c35d
iOS AppRTC: First unit test.
...
Tests basic session ICE connection by stubbing out network components, which have been refactored to faciliate testing.
BUG=3994
R=jiayl@webrtc.org , kjellander@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8002 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 07:21:34 +00:00
pbos@webrtc.org
c37e72e890
Make setting identical RTP extensions a no-op.
...
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.
BUG=1788,4077
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 18:51:13 +00:00
wzh@webrtc.org
433006a6c2
Fixed style issues from lint and got rid of unused fields.
...
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7995 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 17:39:43 +00:00
glaznev@webrtc.org
8390c2762e
Add two unit tests for Android AppRTCDemo.
...
First unit test will create peer connection client, run
for a few second, close it and verify that there were
no any errors and local video was rendered.
Second unit test will run peer connection in a loopback mode.
To run the test from command line install AppRTCDemoTest.apk
and execute the command:
adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7991 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 19:51:12 +00:00
pbos@webrtc.org
896888b7e4
Remove min bitrate from simulcast streams.
...
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/38529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 15:40:56 +00:00
pbos@webrtc.org
9eacb8cc59
Make P2PTestConductor use VirtualSocketServer.
...
Permits running JsepPeerConnectionP2PTestClient in parallel.
TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*
Review URL: https://webrtc-codereview.appspot.com/37459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:03:19 +00:00
pbos@webrtc.org
c62749fb47
Parallelize MediaRecorder unittests.
...
Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.
TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/34589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:01:20 +00:00
jiayl@webrtc.org
27f5317560
Use the prod GAE server in AppRTCDemo for iOS.
...
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-31 00:26:20 +00:00
jiayl@webrtc.org
5eb71eb4f4
Fix style issues from lint.
...
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7984 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 22:44:11 +00:00
glaznev@webrtc.org
b2bda67497
Removing old channel code from a few more places.
...
Plus adding peer connection close event.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7982 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:15:43 +00:00
jiayl@webrtc.org
c5fd66dcdf
Accept incoming pings before remote answer is set to reduce connection latency.
...
BUG=4068
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 19:23:37 +00:00
henrika@webrtc.org
b024da3122
Add support for audio device selection in AppRTCDemo.
...
Summary:
- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.
TBR=glaznev
BUG=4103,4109
Review URL: https://webrtc-codereview.appspot.com/31239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-29 10:35:06 +00:00
pthatcher@webrtc.org
5ad4178137
Move the Jingle-specific network code into webrtc/libjingle.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-23 22:14:15 +00:00
sprang@webrtc.org
46d4d29a75
Add field trial for screenshare bitrates when using temporal layers.
...
BUG=
R=pbos@webrtc.org , pthatcher@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-23 15:19:35 +00:00
braveyao@webrtc.org
086c8d5a02
Use a temporary buffer to scale a screencast in OnFrameCaptured
...
BUG=3903
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/23909005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-22 05:46:42 +00:00
pthatcher@webrtc.org
4c0544ab07
Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository.
...
Also, fix the includes and header guards of examples/call.
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 22:29:55 +00:00
tkchin@webrtc.org
7ce4a584aa
Add initWithCoder to RTCEAGLVideoView.
...
Allows for proper OpenGL initialization if view is created from
storyboard.
BUG=3896
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 20:47:35 +00:00
jiayl@webrtc.org
a6f7ba6848
Add a AppRTCDemo setting to change the GAE server.
...
BUG=4041
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 17:32:14 +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
pthatcher@webrtc.org
5647877b2d
Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 03:32:59 +00:00
pthatcher@webrtc.org
aacc23465b
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
(This is the 3rd try)
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 20:31:29 +00:00
jiayl@webrtc.org
16a05dddb8
Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode.
...
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7955 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 20:12:03 +00:00
pthatcher@webrtc.org
f5847d7746
Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 17:09:11 +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
a9cf079248
Rename external_hmac_ctx_t to ExternalHmacContext.
...
_t types are reserved by POSIX.
R=juberti@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/33699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7944 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 09:12:21 +00:00
pthatcher@webrtc.org
4cb3856a4d
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
...
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.
BUG=
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
This is an un-revert of r7992 and r7993.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 01:22:02 +00:00
pthatcher@webrtc.org
bc03192560
Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 22:15:11 +00:00
tommi@webrtc.org
209df9bf77
Change MockStatsObserver to grab values inside of OnComplete.
...
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 14:09:05 +00:00
pbos@webrtc.org
e728ee03ba
Remove or rename typedefs with _t prefixes.
...
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org , hta@webrtc.org , stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 13:43:55 +00:00
guoweis@webrtc.org
950c518251
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Committed: https://code.google.com/p/webrtc/source/detail?r=7906
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 23:01:31 +00:00
pthatcher@webrtc.org
f050791ba0
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
...
This reverts r7992.
It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:09:08 +00:00
guoweis@webrtc.org
55360ae402
Revert "Add adapter_type into Candidate object."
...
This reverts commit aaf02cc2d4TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 05:28:10 +00:00
guoweis@webrtc.org
aaf02cc2d4
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 23:03:10 +00:00
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
tommi@webrtc.org
e2e199b894
Clean up StatsObserver's OnComplete methods (address TODOs).
...
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
buildbot@webrtc.org
032b802a8c
(Auto)update libjingle 82121498-> 82126219
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf
Remove unneeded ctor and add a more practical one
...
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f
Add thread asserts to StatsCollector.
...
Also adding a "ForTest" postfix to a test-only method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:44:48 +00:00
pbos@webrtc.org
fb108b5a28
Revert r7885.
...
Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org ,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 08:04:50 +00:00
pbos@webrtc.org
18a3896bd2
Revert r7886:7887.
...
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org ,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
magjed@webrtc.org
e575e9c40f
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
...
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
dee76f3b89
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982
Switch kStatsValueName* constants to be enums instead of char*.
...
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 17:41:28 +00:00
pthatcher@webrtc.org
40b276ea7b
Cleanup little things found when refactoring.
...
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/33519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +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
13518951e3
Remove old_factory from WebRtcVideoEngine.
...
Minor pending cleanup.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:14:30 +00:00
perkj@webrtc.org
128fabaf7b
Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
...
Original cl description:
Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 12:25:57 +00:00
buildbot@webrtc.org
a85307737c
(Auto)update libjingle 81702493-> 81755413
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 09:01:18 +00:00
tommi@webrtc.org
aa2c342c10
Add back a constructor to fix FYI build.
...
TBR=perkj
Review URL: https://webrtc-codereview.appspot.com/24349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 20:23:06 +00:00
tkchin@webrtc.org
87776a8935
iAppRTCDemo: WebSocket based signaling.
...
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.
BUG=
R=glaznev@webrtc.org , jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 19:32:35 +00:00
pthatcher@webrtc.org
0babb4a4e6
Fix a comment.
...
R=juberti@webrtc.org , pbos@webrtc.org , sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 19:01:45 +00:00
tommi@webrtc.org
c9d155faeb
Move implementation of types in statstypes. to its cc file.
...
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
...
BUG=4034
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00
tommi@webrtc.org
5c3ee4bce6
Add empty implementation file that will hold statstypes.h implementation.
...
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:47:01 +00:00
glaznev@webrtc.org
eef85387ec
Fix AppRTCDemo closing error for KK and JB Android devices.
...
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 01:29:17 +00:00
andrew@webrtc.org
3b3c406908
Revert 7826 "Change Android PeerConnectionUnittest to build usin..."
...
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575
> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
>
> This also add a new build target to build java PeerConnection using Chromes build macros.
>
> BUG=4031
> R=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28189004
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:21:50 +00:00
perkj@webrtc.org
ed7824b1c0
Change Android PeerConnectionUnittest to build using Chrome macros.
...
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 15:41:01 +00:00
glaznev@webrtc.org
e2a9261f3e
Improve AppRTCDemo connection speed by sending all
...
http POST requests asynchronously.
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 20:11:06 +00:00
kjellander@webrtc.org
bd8cc0b914
Add codereview.settings to the /talk subdirectory
...
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 13:47:37 +00:00
kjellander@webrtc.org
599e299b9d
cricket::VideoFrame int64 to int64_t.
...
Needed for successful compile of ios arm64.
BUG=3898
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30359004
Patch from Zeke Chin <tkchin@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 09:42:57 +00:00
bemasc@webrtc.org
9b5467e88d
Fix assertion failure when closing data channel, and add a unit test.
...
BUG=4066
R=jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7816 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 23:16:52 +00:00
glaznev@webrtc.org
4b407aa985
Update AppRTCDemo README with information on 3-dot-apprtc server
...
and new command line arguments.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 22:42:59 +00:00
guoweis@webrtc.org
7169afd9d5
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
...
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:59:29 +00:00
glaznev@webrtc.org
369746bcb8
Support new WebSocket signaling format.
...
- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.
BUG=3937,3995,4041
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:28:52 +00:00
pbos@webrtc.org
0fb6ad2004
Check if cpu_monitor_ exists before Stop().
...
R=asapersson@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 13:44:29 +00:00
asapersson@webrtc.org
d8aed6b321
Verify that cpu_monitor exists before calling Stop().
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 12:37:47 +00:00
pthatcher@webrtc.org
eb0954248d
Don't reset sequence number for a stream on deactivate/reactivate.
...
BUG=chromium:431908
R=pbos@webrtc.org , sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 00:34:10 +00:00
glaznev@webrtc.org
d01955179a
Change minimum video encoder initialization resolution to
...
176x144 to ensure HW encoder can be initialized.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 23:41:18 +00:00
perkj@webrtc.org
beee9cec22
Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video.
...
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.
TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-02 14:38:18 +00:00
pthatcher@webrtc.org
146e0fd30f
Fix the build by putting in a typecast to avoid a comparison between
...
signed and unsigned ints introduced in cl/81073932.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7776 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 20:07:52 +00:00
glaznev@webrtc.org
dea5173edf
Add start bitrate and vp8 hw acceleration option to
...
Android AppRTCDemo.
- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.
BUG=4046
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 20:02:13 +00:00
buildbot@webrtc.org
32ec0dd032
(Auto)update libjingle 81063831-> 81073932
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-01 17:57:36 +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
tommi@webrtc.org
2c13f659c7
Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-28 10:37:31 +00:00
tkchin@webrtc.org
3e9ad26112
Refactor iOS AppRTC parsing code.
...
Moved parsing code to JSON categories for the relevant objects.
No longer prefer ISAC as audio codec.
BUG=
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31989005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 00:52:38 +00:00
sprang@webrtc.org
a71bb6033b
Revert 7750 "Don't reset sequence number for a stream on deactiv..."
...
> Don't reset sequence number for a stream on deactivate/reactivate.
>
> BUG=chromium:431908
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/32199004
TBR=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7752 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 19:33:15 +00:00
sprang@webrtc.org
31f7a0e710
Don't reset sequence number for a stream on deactivate/reactivate.
...
BUG=chromium:431908
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7750 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 16:55:52 +00:00
perkj@webrtc.org
2faf7eea6f
Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection.""
...
This reverts commit 308e7ff613R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 07:35:37 +00:00
glaznev@webrtc.org
58edb83fd4
Add video encoder fps and bitrate statistics to
...
Android AppRTCDemo UI.
BUG=4045
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 00:39:42 +00:00
