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Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
stefan@webrtc.org
3dd33a6787 Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes. 
BUG=crbug:425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 09:12:23 +00:00
henrik.lundin@webrtc.org
fbd37bd737 Make iSAC SWB own its decoder 
A bug in the ACM codec database caused iSAC-swb to behave differently
from iSAC-wb and -fb. With this fix, all iSAC codecs behave the same
with respect to decoder ownership.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 08:16:29 +00:00
andrew@webrtc.org
e65d9d974c Fix an unitialized variable warning. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35819004

Patch from Sebastien Marchand <sebmarchand@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 22:05:12 +00:00
kjellander@webrtc.org
c429b824b3 GN: Prepare to remove webrtc_base target 
Keep the webrtc_base target temporarily while waiting for
Chromium to pick up this revision. Then we'll update Chromium
and remove the webrtc_base target for real.

This should have been a part of https://code.google.com/p/webrtc/source/detail?r=7140

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 20:22:33 +00:00
aluebs@webrtc.org
c78d81ae89 Re-land "Support 48kHz in AEC" 
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 19:10:55 +00:00
aluebs@webrtc.org
e81c5d6d7e Fix TransientDetectorTest in modules_unittests on Android ARM64 
BUG=webrtc:4200
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 18:01:28 +00:00
minyue@webrtc.org
11af039590 Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64. 
BUG=4199
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 14:22:39 +00:00
asapersson@webrtc.org
df7b65ba01 Change CreateOrGetReportBlockInformation to have one return path. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 13:07:04 +00:00
asapersson@webrtc.org
9ffd8fe96b Indentation changes. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 08:22:50 +00:00
minyue@webrtc.org
7dba7860c7 Setting Opus target application. 
This CL is to allow to set Opus target application at the creation of an encoder.

According to Opus spec, there are three applications:

OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY

BUG=
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 16:01:50 +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
tina.legrand@webrtc.org
ee0c100d54 Revert 8080 "Support 48kHz in AEC" 
> Support 48kHz in AEC
> 
> Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
> Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
> 
> BUG=webrtc:3146
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28319004

TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 10:22:49 +00:00
changbin.shao@intel.com
9691b36995 Cleanup for Rtp Rtcp API test. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 05:42:52 +00:00
stefan@webrtc.org
474e36e623 Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. 
The previous CL was reverted for two reasons:
- Added a static initializer because std::string.
- Landed before the corresponding chromium CL, which has now been landed.

BUG=crbug:425925
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 15:44:47 +00:00
kjellander@webrtc.org
a32d15448d Disable tests failing on Android ARM64 (Nexus9). 
BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.

Review URL: https://webrtc-codereview.appspot.com/33919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 12:46:01 +00:00
stefan@webrtc.org
a1aea10af2 Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps." 
TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-16 13:52:52 +00:00
andresp@webrtc.org
4ba1e44ff0 Remove unnecessary remote bitrate estimator build rule which serves no purpose. 
BUG=4185
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-16 07:50:17 +00:00
aluebs@webrtc.org
64d3c4b9ac Support 48kHz in AEC 
Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 19:52:05 +00:00
aluebs@webrtc.org
d82f55d2a7 Only adapt AGC when the desired signal is present 
Take the 50% quantile of the mask and compare it to certain threshold to determine if the desired signal is present. A hold is applied to avoid fast switching between states.
is_signal_present_ has been plotted and looks as expected. The AGC adaptation sounds promising, specially for the cases when the speaker fades in and out from the beam direction.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 18:07:21 +00:00
stefan@webrtc.org
3e42a8a56a Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. 
BUG=crbug:425925
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 14:45:27 +00:00
henrik.lundin@webrtc.org
1f67b53c88 Remove dual stream functionality in ACM 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 09:36:30 +00:00
asapersson@webrtc.org
0800db74b9 Add percentage of fec packets and recovered media packets to histogram stats: 
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"

BUG=crbug/419657
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 07:40:20 +00:00
andrew@webrtc.org
6c3855258d Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version. 
This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.

Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/36689004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 02:56:06 +00:00
mgraczyk@chromium.org
5a92b78e86 Add beamforming to audioproc_float utility. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 01:28:36 +00:00
andrew@webrtc.org
6b6301588e Move ring_buffer to common_audio. 
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.

R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 00:09:53 +00:00
kjellander@webrtc.org
693e01c910 Fix searching for DirectX SDK during GN build. 
Before that GN just checked for DXSDK_DIR environment variable.
GYP does more and checks registry, let's do the same in GN.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37599004

Patch from Vyacheslav Chigrin <vchigrin@yandex-team.ru>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 21:25:25 +00:00
stefan@webrtc.org
387841ac5c Improved fairness simulation by starting the flows 20 seconds apart. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:45:29 +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
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
mgraczyk@chromium.org
d6e84d9d13 Always copy processed audio to output buffer in ProcessStream. 
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.

R=andrew@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08

Review URL: https://webrtc-codereview.appspot.com/41459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 01:33:54 +00:00
pbos@webrtc.org
0b0c24177b Only return Rtx mode in RTXSendStatus(). 
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.

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

Review URL: https://webrtc-codereview.appspot.com/38569004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 14:15:15 +00:00