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Author SHA1 Message Date
henrik.lundin@webrtc.org
4aecd008dd Add support for 40 and 60 ms frames to AudioEncoderIlbc 
BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37789004

Cr-Commit-Position: refs/heads/master@{#8182}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 13:16:44 +00:00
sprang@webrtc.org
2a6558c2a5 Make sure ByteReader<T>::Read* is properly constified. 
Also, start using it in real code...

BUG=
R=holmer@google.com, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37809004

Cr-Commit-Position: refs/heads/master@{#8181}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 12:38:16 +00:00
kjellander@webrtc.org
7aef80c6d1 GN: Remove webrtc_base target in favor for rtc_base. 
The last reference to the old target name was
removed in https://crrev.com/7c9149860a8a0ca24350d2e80dbc280990a0cbb7

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33079004

Cr-Commit-Position: refs/heads/master@{#8179}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-28 07:55:45 +00:00
marpan@webrtc.org
9b64a6edd7 Adjust parameter in videoprocessor_integrationtest for VP9. 
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/35919004

Cr-Commit-Position: refs/heads/master@{#8178}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 23:59:16 +00:00
marpan@webrtc.org
dc8a9da386 Adjust qp-max settinhg in VP9 wrapper. 
More closely matches the qp-max setting used in VP8.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39709004

Cr-Commit-Position: refs/heads/master@{#8177}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 23:08:39 +00:00
andrew@webrtc.org
922cfcd150 Use non-zero data in AudioRingBufferTest. 
TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35909004

Cr-Commit-Position: refs/heads/master@{#8176}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 21:59:44 +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
henrik.lundin@webrtc.org
8bb32d600b Minor updates to AudioEncoderCng 
Removing sample_rate_hz_ from AudioEncoderCng and from the config
struct. The sample rate will now be read from the underlying speech
codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40559004

Cr-Commit-Position: refs/heads/master@{#8173}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 20:54:22 +00:00
tnakamura@webrtc.org
db1ebf6c0c Add jakehilton@gmail.com to AUTHORS 
BUG=3918
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34889004

Cr-Commit-Position: refs/heads/master@{#8172}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 19:15:48 +00:00
henrik.lundin@webrtc.org
478cedc055 Add new methods to AudioEncoder interface 
The following three methods are added:
rtp_timestamp_rate_hz()
SetTargetBitrate()
SetProjectedPacketLossRate()

Default implementations are provided, and a few overrides are
implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new
methods to the underlying speech codec.

BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34049004

Cr-Commit-Position: refs/heads/master@{#8171}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 18:25:40 +00:00
bjornv@webrtc.org
5614cf16e7 audio_processing: Use fixed aggregation window in delay metrics 
Previously, the delay estimate history was reset every time the metrics were pulled. This required all clients to be on the same thread and make use of one call.

Now we use a fixed aggregation window of one second and when a client pulls the metrics you get the latest value.
Under certain circumstances like tests you would like to have the aggregation window set to the recording length. We therefore turn on the fixed aggregation window after the first call.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38759004

Cr-Commit-Position: refs/heads/master@{#8170}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 18:10:27 +00:00
kjellander@webrtc.org
6e251822cd Whitespace change after enabling gnumbd 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/37019004

Cr-Commit-Position: refs/heads/master@{#8169}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 16:46:13 +00:00
kjellander@webrtc.org
ccd608eeab Whitespace change for git updater 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/41669004

Cr-Commit-Position: refs/heads/master@{#8168}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 16:37:48 +00:00
kjellander@webrtc.org
0bc73a1b72 Whitespace change to trigger git updater 
TBR=machenbach@chromium.org

Review URL: https://webrtc-codereview.appspot.com/34869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 14:13:13 +00:00
kjellander@webrtc.org
f68ffca050 Add PRESUBMIT check for GYP files including source files above itself. 
This is needed because some tools does not support files
located above the project generated.

BUG=4185
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 13:13:24 +00:00
kjellander@webrtc.org
76e5e207ad Roll chromium_revision 4664fe0..9070a80 (312733:313233) 
Relevant changes:
* src/third_party/boringssl/src: 5fa3eba..347f025
* src/third_party/libvpx: 8dc6ea9..5da40ca
* src/tools/gyp: adb7d24..b28bd7d
* src/tools/swarming_client: e98dde9..d863df3
Details: 4664fe0..9070a80/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 13:11:10 +00:00
asapersson@webrtc.org
273fbbb921 Update StreamDataCounter with FEC bytes. 
Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"

Correct media payload bytes in StreamDataCounter to not include FEC bytes.

Fix stats for rtcp packets sent/received per minute (regression from r7910).

BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 12:17:29 +00:00
bjornv@webrtc.org
70117a83d4 AEC: Implements a new function for calculating delay metrics 
Two new member variables have been added and the code for calculating the delay metrics have been moved to a function.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-27 11:30:54 +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
pkasting@chromium.org
df9a41d270 Fix bug in GetREDStatus(): it doesn't actually return the current status. 
BUG=none
TEST=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 22:35:29 +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
andrew@webrtc.org
041035b390 Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface. 
Integrate it in Blocker to demonstrate use.

TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com

Review URL: https://webrtc-codereview.appspot.com/36799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 21:23:53 +00:00
pkasting@chromium.org
4dba2e98a2 Consolidate anonymous namespace content and file-static methods to all be in the 
anonymous namespace, in preparation for refactoring a few of the functions a
little.

No code change.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 19:59:32 +00:00
kjellander@webrtc.org
d7e34e1086 Make it easier to use external libyuv + cleanup GYP files. 
It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 19:17:26 +00:00
bjornv@webrtc.org
d25c034051 Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16() 
BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:32:47 +00:00
tommi@webrtc.org
04cd466bd5 Move ThreadChecker into rtc_base_approved. 
To do this, I'm removing ThreadChecker's dependency on the 'Thread' class, so that the checker works with any thread and doesn't rely on TLS.
Also simplifying CriticalSection's implementation on Windows since a critical section on Windows already knows what thread currently owns the lock.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:27:29 +00:00
marpan@webrtc.org
38d11b8529 Enable encoder multi-threading for VP9. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 15:21:36 +00:00
kwiberg@webrtc.org
6f200b5b87 Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h") 
Some out-of-tree code that uses base/scoped_ptr.h is defining nullptr
to 0, which causes an obvious compilation error and perhaps other
subtle problems. I'm hoping to get that sorted out and re-land this CL
soon.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 13:03:32 +00:00
henrik.lundin@webrtc.org
b6fab2b1cd Introduce rtc::CheckedDivExact 
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.

COAUTHOR:kwiberg@webrtc.org

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 11:08:53 +00:00
kwiberg@webrtc.org
19eb4e4b86 Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h 
The latter file was more up-to-date. The files are now identical
with the following exceptions:

  * The namespace used (rtc vs. webrtc).

  * The name of the include guard.

  * base/scoped_ptr.h still has two extra methods, accept() and use().

  * base/scoped_ptr.h still includes webrtc/base/common.h even though
    it doesn't need it itself, since several .cc files expect to get
    it for free by incuding base/scoped_ptr.h. This is of course bad
    manners, and the "unused" include will be removed in a future CL.

A later CL will remove system_wrappers/interface/scoped_ptr.h.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-26 08:57:57 +00:00
kjellander@webrtc.org
995b4c9e8a Remove win_asan trybot from PRESUBMIT.py 
Removing it since it no longer exists.
See https://codereview.chromium.org/872263002/

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-25 19:27:03 +00:00
kjellander@webrtc.org
acb8085678 Roll chromium_revision c086b4e..4664fe0 (312108:312733) 
Mainly to pick up the MIPS changes in
https://codereview.chromium.org/843563002/
for which the changes in
https://webrtc-codereview.appspot.com/41399004/
are included in this CL.

Relevant changes:
* src/third_party/android_tools: 56b3d3e..aaeda3d
* src/third_party/boringssl/src: ca9a538..5fa3eba
* src/third_party/libvpx: 4f9bd1b..8dc6ea9
* src/third_party/openmax_dl: 1a4171c..8f7bf0b
* src/tools/gyp: 194ec65..adb7d24
* src/tools/swarming_client: 0a795bd..e98dde9
Details: c086b4e..4664fe0/DEPS

Clang version was not updated in this roll.

BUG=4214, 4222
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-25 19:17:56 +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
fdegans@chromium.org
8919cfe9ce Change a GYP reference to cpufeatures.gypi 
This will allow us to move the remaining GYP file in android_tools
to the chromium repository by removing the direct reference to it.

BUG=webrtc:4115
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 16:35:17 +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
minyue@webrtc.org
456f01441a Re-allowing RED in voice engine. 
Path of audio RED packets was blocked in r4692 by accident. It ought be enabled again.

BUG=3619
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 11:58:42 +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
stefan@webrtc.org
73ee4537be Switch to use range based loops in the BWE simulation framework. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-23 08:29:52 +00:00
davidben@webrtc.org
36d5c3cb44 Leave BIO_METHOD non-const. 
This breaks building against OpenSSL upstream, which is still supported on iOS.
This reverts part of https://webrtc-codereview.appspot.com/34649004.

BUG=none
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 23:06:17 +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
jiayl@webrtc.org
7e5b380437 Fix a crash in AllocationSequence. 
Internal bug 19074679.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8130 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 21:28:39 +00:00
kjellander@webrtc.org
ff108fe508 Revert 8125 "Modify some tests to never use DTX disable mode" 
Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293

Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
        ^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
        ^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
        ^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
        ^
2 errors generated.

> Modify some tests to never use DTX disable mode
> 
> DTX disable mode will be removed as a part of the ACM redesign work.
> 
> COAUTHOR:kwiberg@webrtc.org
> 
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/34769004

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 19:02:03 +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
jlmiller@webrtc.org
ea1c84285c Correct GetDriveType error handling. 
BUG=4020
R=brucedawson@google.com, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 17:44:19 +00:00
henrik.lundin@webrtc.org
043db24767 Modify some tests to never use DTX disable mode 
DTX disable mode will be removed as a part of the ACM redesign work.

COAUTHOR:kwiberg@webrtc.org

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 13:30:58 +00:00
stefan@webrtc.org
e5251ad63c Integrate send-side BWE into simulation framework. 
BUG=4173
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 10:10:53 +00:00