cc476aa038
Fix a name collision with Android libc++
...
The Android libc++ has a symbol called '_P'
This CL renames a property called _P in webrtc.
BUG=chromium:427718
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30009004
Patch from Fabrice de Gans-Riberi <fdegans@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7579 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:01:25 +00:00
b7ed7799e7
Implement conference-mode temporal-layer screencast.
...
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 13:08:10 +00:00
3bf3d238c8
Configure A/V sync in WebRtcVideoEngine2.
...
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/23249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 12:59:34 +00:00
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
8328e7c44d
Revert "Revert part of r7561, "Refactor audio conversion functions.""
...
This restores the conversion changes to AudioProcessing originally
added in r7561, with minor alterations to ensure it passes all tests.
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/28899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7574 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 04:58:14 +00:00
14146e40aa
arm64 iOS build.
...
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.
BUG=3898
R=kjellander@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 00:14:39 +00:00
d0cf68ee37
Add 15 fps support for Android devices with missing 15 fps
...
camera mode.
Some latest Android devices support only 30 fps for front camera,
but HW VP8 encoder performance is not enough for 720p 30 fps
encoding. Add 15 fps support for these devices by allowing
frame drop in Android camera wrapper.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7571 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 18:38:26 +00:00
8aa4d2d2cd
Creating a C++ wrapper class for VAD
...
Also adding a mock. This work is part of an ongoing effort to
encapsulate encoders in AudioEncoder classes. The CNG encoder will also
be implemented as an AudioEncoder class, and will also contain a VAD
C++ wrapper.
BUG=3926
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7570 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 13:23:25 +00:00
bcfb4d0403
Revert part of r7561, "Refactor audio conversion functions."
...
Specifically, revert this part:
"Remove hacks in AudioBuffer intended to maintain bit-exactness with
the float path. The conversions etc. are now all natural, and
instead we enforce close but not bit-exact output between the two
paths."
But keep the conversion function rename, since that doesn't seem to be
causing problems.
R=tina.legrand@webrtc.org , bjornv@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7569 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 11:16:06 +00:00
4fc4addc81
Refactor audio conversion functions.
...
Use a consistent naming scheme that can be understood at the callsite
without having to refer to documentation.
Remove hacks in AudioBuffer intended to maintain bit-exactness with the
float path. The conversions etc. are now all natural, and instead we
enforce close but not bit-exact output between the two paths.
Output of ApmTest.Process:
https://paste.googleplex.com/5931055831842816
R=aluebs@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7561 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-30 03:40:10 +00:00
776e6f289c
Use external VideoDecoders in VideoReceiveStream.
...
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org , stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 15:28:39 +00:00
2dd3134e50
Add stats for duplicate sent and received NACK requests.
...
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 12:42:30 +00:00
f567095f62
common_audio: Removed macro WEBRTC_SPL_RSHIFT_W32
...
Replaces the trivial macro WEBRTC_SPL_RSHIFT_W32 with >> at various places in common_audio and removes it.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7558 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 10:29:16 +00:00
7f10513efc
Remove unused code in overuse detector.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7557 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 10:05:21 +00:00
decd9306ae
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
...
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.
BUG=3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:38:50 +00:00
cfe3845b66
Enable G.722 for Chromium builds
...
BUG=3909
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7555 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:32:44 +00:00
663fdd02fd
Make an AudioEncoder subclass for Opus
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 07:28:36 +00:00
ffeaeed8c1
Make NSinst_t* const and rename to self in ns_core
...
This is only to make the code more readable and maintainable.
It generates a bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7550 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:52:09 +00:00
269fb4bc90
move xmpp and p2p to webrtc
...
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
8b1b23f8f8
Make local functions static and dropWebRtcNs_ in ns_core
...
This is only to make the code more readable and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7548 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 21:06:57 +00:00
28b54671cb
Make all comments whole sentences in ns_core
...
This is done to make the code more readable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7547 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 20:56:53 +00:00
bd6bdca57f
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7546 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 18:06:42 +00:00
a296725d0e
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:05:43 +00:00
67ca26e087
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
...
The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.
Affected components:
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:03:10 +00:00
ff8a98e352
Use neteq_unittest_tools in audio_decoder_unittests
...
With the recent move of RtpFileReader to the rtp_test_utils target
(in r7536), it is now possible to let audio_decoder_unittests depend
on neteq_unittest_tools without breaking the Android build.
BUG=2692
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7542 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 09:47:13 +00:00
820efd5b55
Fix double backslashes in incoming_video_stream.cc
...
Originally uploaded in https://codereview.appspot.com/149160043/ .
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7541 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 08:47:16 +00:00
aada86b261
Add a simple AudioConverter class.
...
This will be used to refactor AudioProcessing/AudioBuffer. We can
enable alternate downmixing schemes in AudioProcessing by pulling
the conversion logic out of AudioBuffer.
The unit test is largely stolen from voice_engine/utility_unittest.cc.
As commented, the voice_engine routines should be replaced with
AudioConverter.
BUG=chromium:405270
R=aluebs@webrtc.org , mgraczyk@chromium.org
TBR=kwiberg
Review URL: https://webrtc-codereview.appspot.com/30779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7538 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:18:17 +00:00
33a0e2d7ef
Only configure the SSL library in one place.
...
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.
This is to avoid colliding with Chromium's transition away from NSS.
This is a fixup of https://webrtc-codereview.appspot.com/29559004 to avoid
breaking use_legacy_ssl_defaults.
BUG=chromium:413497
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7537 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:13:40 +00:00
aca5803b19
Move (test) RtpFileReader to a lightweight target.
...
Moves RtpFileReader to rtp_packet_parser and renames it to
rtp_test_utils which is allowed to rely on rtp_rtcp.
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/24979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7536 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 18:01:03 +00:00
b787f4c593
Move scoped_ptr "free" functions into the webrtc namespace.
...
Resolves a conflict with Chromium's scoped_ptr on the recently added
make_scoped_ptr().
TEST=local Chromium Linux build passes.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7535 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 17:42:22 +00:00
df429882af
Upgrade our scoped_ptr copy to match Chromium's latest.
...
In particular add the move constructor and assignment operator.
Diff between our version and Chromium's:
https://paste.googleplex.com/4887047529562112
R=henrike@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7531 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 16:12:38 +00:00
a37f1dd6b8
Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile
...
This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 12:58:18 +00:00
0552356fda
isacfix: Refactor big-endian reading and writing
...
Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 11:25:37 +00:00
9fed099208
Increase max trace message size to 1024 characters.
...
A recent CL by pbos:
https://code.google.com/p/webrtc/source/detail?r=7518
added long log messages and triggered errors on the DrMemory bot due to
WEBRTC_TRACE. The trace mechanism _should_ truncate the log strings
but something appears to be going awry.
This sweeps the problem under the rug, but given that WEBRTC_TRACE
should die fairly soon, seems to be a reasonable tradeoff.
TEST=passing try on DrMemory.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27849004
Patch from Andrew MacDonald <andrew@webrtc.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 09:31:05 +00:00
c86ec3e3bc
Fix ::~LogMessage to print as a string.
...
R=andrew@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/26949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7525 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 09:22:03 +00:00
39b1743116
Adding the subtool rtcBot report visualizer
...
This tool for visualize the output reports of rtcBot by calculating
the average and max of a specific stats and plot the output.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7519 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 09:26:16 +00:00
ad3b5a5c16
Move min transmit bitrate to VideoEncoderConfig.
...
min_transmit_bitrate_bps needs to be reconfigurable during a call (since
this is currently set only for screensharing through libjingle and can't
be set once and for all for the entire Call.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/28779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7518 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-24 09:23:21 +00:00
7e19a11a71
Break out WebRtcNs_ComputeDdUpdate function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7514 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:54:33 +00:00
f8ea0d5518
Break out WebRtcNs_UpdateNoise function in ns_core
...
This is done in order to make the code more readible and maintainable.
It generates bit-exact output.
BUG=webrtc:3811
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7513 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:49:42 +00:00
799e88ae19
Break out FFT function in ns_core
...
This is done in order to make the code more readible and maintainable.
This introduces an error of only +1 and -1.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:36:42 +00:00
8454ad88ed
Break out ComputeSnr function in ns_core
...
This is done in order to make the code more readible and maintainable.
The output is bit-exact.
BUG=webrtc:3811
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7511 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 19:34:14 +00:00
0d3e254c89
Adding three video conference bots test
...
A video conference between three bots, each bot creating two
peerConnections, and each peer connected to one of the other bots.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7510 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 16:45:07 +00:00
0e19d0c2aa
Adding file from test.webrtc.org domain to be downloaded
...
This has been configured to allow cross domain to access this generated
file:
https://test.webrtc.org/test-download-file/9000KB.data
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7509 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 15:41:30 +00:00
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
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.
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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
2192701135
Using the Unused turn configuration in two way test
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R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7504 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:40:53 +00:00
ad553a2731
Let video_loopback use internal VCM capturers.
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R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7503 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 08:24:02 +00:00
fce8f5d319
NOTE: This code review based on the running issue:
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https://webrtc-codereview.appspot.com/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:24:20 +00:00
3382059e55
Adding Two way video and audio streaming test to RtcBot
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NOTE: This code review based on this running issue:
https://review.webrtc.org/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7498 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 17:17:15 +00:00
e9b7d03db6
HTTPS Server used instead of HTTP for loading the bots to avoid the media permission pop-up clicks every time running the test.
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This code review based on the running issue:
https://webrtc-codereview.appspot.com/24939004/
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 16:34:25 +00:00
