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Author SHA1 Message Date
minyue@webrtc.org
2dc6f3154d Adapting bitrate according to maxplaybackrate for Opus. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 05:33:10 +00:00
andrew@webrtc.org
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
tkchin@webrtc.org
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
jiayl@webrtc.org
50ca986bc1 Improve the logging when a TCP connection is deleted. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 23:50:54 +00:00
glaznev@webrtc.org
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
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
minyue@webrtc.org
8219529b98 Cleaning up r7562-7567. 
Wrongly used git svn dcommit for committing a CL.

Then two reverts were applied.

Still something needs to be cleaned.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 08:23:54 +00:00
buildbot@webrtc.org
879fac81d1 (Auto)update libjingle 78822708-> 78823675 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:50:13 +00:00
minyue@webrtc.org
5f73a37597 Revert 7563 "before rebase" due to wrong submission 
> before rebase

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:49:58 +00:00
minyue@webrtc.org
c11cc8d947 Revert 7564 "to submit" due to wrong submission 
> to submit

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:46:47 +00:00
minyue@webrtc.org
de386bf67b to submit 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:20:09 +00:00
minyue@webrtc.org
c673bb9f29 before rebase 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:57 +00:00
minyue@webrtc.org
0b62672576 adding default rates 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:49 +00:00
andrew@webrtc.org
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
pbos@webrtc.org
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
asapersson@webrtc.org
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
bjornv@webrtc.org
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
asapersson@webrtc.org
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
kwiberg@webrtc.org
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
henrik.lundin@webrtc.org
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
buildbot@webrtc.org
1abc146aa5 (Auto)update libjingle 78738075-> 78738103 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:14:14 +00:00
perkj@webrtc.org
7998089789 ApprtDemo Android: Switch between front and back camera. 
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.

BUG=3786
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:10:03 +00:00
kwiberg@webrtc.org
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
minyue@webrtc.org
2623695dfb Renaming bandwidth to bitrate in webrtcvoiceengine. 
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.

This is to remove the confusion inside webrtcvoiceengine

BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 02:27:08 +00:00
aluebs@webrtc.org
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
henrike@webrtc.org
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
aluebs@webrtc.org
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
aluebs@webrtc.org
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
henrike@webrtc.org
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
buildbot@webrtc.org
ae694effd8 (Auto)update libjingle 78642371-> 78680406 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 17:37:17 +00:00
bjornv@webrtc.org
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
bjornv@webrtc.org
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
henrik.lundin@webrtc.org
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
perkj@webrtc.org
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
buildbot@webrtc.org
fbd55cb27d (Auto)update libjingle 78616359-> 78642371 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 05:35:35 +00:00
tommi@webrtc.org
f15dee6980 Check if a datachannel in the current local description is an sctp channel before assuming rtp. 
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 22:15:04 +00:00
andrew@webrtc.org
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
henrike@webrtc.org
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
pbos@webrtc.org
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
andrew@webrtc.org
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
glaznev@webrtc.org
243eb8e9af Adding setting screen to AppRTCDemo. 
- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.

BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 17:22:15 +00:00
buildbot@webrtc.org
068b529f46 (Auto)update libjingle 78583324-> 78583691 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 16:20:42 +00:00
andrew@webrtc.org
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
pthatcher@webrtc.org
2e7ee4b28b Fix the SrtpFilter crash caused by two local offers. 
BUG=http://crbug.com/421774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 16:10:29 +00:00
pbos@webrtc.org
efc82c2c73 Implement screencast settings for WebRtcVideoEngine2. 
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 13:58:00 +00:00
henrik.lundin@webrtc.org
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
kwiberg@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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