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Author SHA1 Message Date
guoweis@webrtc.org
4bbd3c83a8 fix a bug in the logic when new Networks are merged. This happens when 
we have 2 networks with the same key

BUG=410554 in chromium

http://code.google.com/p/chromium/issues/detail?id=410554

Corresponding change in chromium is
https://codereview.chromium.org/536133003/

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 13:54:45 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h. 
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 10:40:56 +00:00
andrew@webrtc.org
641bda6f9c Initialize ChannelBuffer's memory to avoid uninitialized reads. 
Removed the zero out memset in this change:
https://review.webrtc.org/24469004/

assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.

TBR=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/28429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 23:11:44 +00:00
brettw@chromium.org
519c9e207d Convert GN visibility to be a list. 
GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:45:18 +00:00
andrew@webrtc.org
17454f79dc Add ctors to ChannelBuffer to enable copying on construction. 
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.

R=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/24469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:27:04 +00:00
henrik.lundin@webrtc.org
c64246f42c Set a default speech type in iSAC wrapper 
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.

BUG=crbug/411162
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:40:58 +00:00
henrik.lundin@webrtc.org
ed8bcd3ac5 Starting to implement the new ACM API 
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:13:19 +00:00
houssainy@google.com
9600519147 Adding the ability to test on Chrome for Android. 
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:01:40 +00:00
bjornv@webrtc.org
37c39f3784 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16 
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.

Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:21:56 +00:00
bjornv@webrtc.org
0d394f3609 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16 
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:19:39 +00:00
houssainy@google.com
c77e4d6aef - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. 
- Select BotType using nodeJs terminal command.

- ping_pong.js test added.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 10:36:11 +00:00
stefan@webrtc.org
fe16167507 Fix RTT calculations for send-only channels. 
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:45:25 +00:00
sprang@webrtc.org
c30e9e2300 Ignore FEC packet in stats, if it is first packet on ssrc. 
BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:20:18 +00:00
kjellander@webrtc.org
6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_" 
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
f68cf93e1b Add video_capture_tests_apk_target 
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.

BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:35:51 +00:00
kjellander@webrtc.org
a781f68712 Fix rm command for class cleanup in r7091 
In https://webrtc-codereview.appspot.com/20339004
the rm command was missing 'r' for recursive mode.

TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 22:11:28 +00:00
kjellander@webrtc.org
9510022e1f Cleanup temporary class files for OpenSlDemo 
I've seen tryjobs failing when they shouldn't on
the Android trybots and I suspect this might have
something to do with it.

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 18:03:45 +00:00
henrik.lundin@webrtc.org
8f073c5054 Create a new interface for AudioCodingModule 
This is a first draft of the interface, and is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:16:23 +00:00
kjellander@webrtc.org
fed47dc205 Drop buildbot_tests.py script 
This is no longer used since the buildbots have moved
over to recipes (where these arguments are configured).
See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73
for details.

This is essentially a revert of
https://webrtc-codereview.appspot.com/1021006

BUG=None
TESTED=None
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 08:25:38 +00:00
henrik.lundin@webrtc.org
bcf75e32a3 Modifying audio_coding/codecs/OWNERS 
Adding myself.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 07:18:50 +00:00
bjornv@webrtc.org
c2c4117477 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing 
Affected components:
* AECMobile
  - Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 06:01:53 +00:00
henrike@webrtc.org
f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error 
STUN_ERROR_ALLOCATION_MISMATCH."

TBR=jiayl@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/15359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 22:21:33 +00:00
jiayl@webrtc.org
574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:11:34 +00:00
aluebs@webrtc.org
021e76fd39 Add support for WAV output in audioproc 
The default output is a WAV file, except if the --pcm_output flag is set.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 18:12:00 +00:00
brettw@chromium.org
afa77cd803 Add direct_dependent_config to desktop_capture in GN build. 
This allows us to remove some configs in the Chrome build that should come
automatically from depending on this target.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 17:00:55 +00:00
andresp@webrtc.org
fa822b940f Fix strange owners files with comments that crashs "git cl presubmit" 
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 14:25:07 +00:00
kjellander@webrtc.org
79ee97cf43 [MIPS] Fix gn gen failure for MIPS in webrtc 
Fixes the following failure for mips:
"ERROR at //third_party/webrtc/BUILD.gn:136:7: Undefined variable for +=.
      cflags += [ "-mhard-float" ]
      ^-----
I don't have something with this name in scope now."

BUG=3441
TEST=In Chromium. Passing compile locally on Linux using:
gn gen out-gn/mips --args="is_debug=false os=\"android\" cpu_arch=\"mipsel\"" --verbose &&  ninja -C out-gn/mips all
gn gen out-gn/arm --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" --verbose &&  ninja -C out-gn/arm all
gn gen out-gn/x86-linux --args="is_debug=false os=\"linux\"" --verbose &&  ninja -C out-gn/x86-linux webrtc

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15349004

Patch from Gordana Cmiljanovic <Gordana.Cmiljanovic@imgtec.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 14:10:49 +00:00
houssainy@google.com
38ef664418 Moving the api.js and bot.js to /rtcbot/bot/ to be shared between 
/borwser and /android

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:44:47 +00:00
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files. 
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:28:48 +00:00
bjornv@webrtc.org
3cbd6c26c8 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. 
This changes some method signatures to better reflect how callers are actually
using them.  This also has the tendency to make signatures more consistent about
e.g. using int (instead of int16_t) for lengths of things like vectors, and
using int16_t (instead of int) for e.g. counts of bits in a value.

This also removes a couple of functions that were only called in unittests.

BUG=3353,chromium:81439
TEST=none
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:21:44 +00:00
henrik.lundin@webrtc.org
f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo 
The name of the copy method was confusing. This change makes the
code easier to read where the method is used.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 10:58:43 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths. 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 09:55:40 +00:00
kwiberg@webrtc.org
51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_ 
R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 08:42:44 +00:00
henrik.lundin@webrtc.org
7825b1abf9 Add support for multi-channel DTMF tone generation 
This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:39:21 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:07:44 +00:00
stefan@webrtc.org
01581da711 Fix audio/video sync when FEC is enabled. 
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.

BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 06:48:14 +00:00
andresp@webrtc.org
bfd7a8c448 Fix compile errors on webrtc/base. 
R=fbarchard@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:59:52 +00:00
andresp@webrtc.org
0229cbae33 Remove ambiguous call to MakeCheckOpString. 
BUG=3777
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:53:29 +00:00
fbarchard@google.com
9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error 
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 23:05:07 +00:00
tkchin@webrtc.org
5b83af49c1 Fix leak of NSAutoreleasePool. 
This looks like something that's no longer applicable. From what I saw this code path isn't on a static initializer that runs before main. Should be okay to drain (release) pool outside of this scope.

BUG=3659
R=henrike@webrtc.org, jiayl@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 22:53:34 +00:00
henrike@webrtc.org
1b8b4c4959 Revert 7041 " Audio codecs to include webrtc/typedefs.h" 
Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio

R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/19219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 19:42:16 +00:00
pbos@webrtc.org
26c0c41a06 Network up/down signaling in Call. 
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:17:12 +00:00
andresp@webrtc.org
9730d3aae9 Audio codecs to include webrtc/typedefs.h 
Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 14:37:18 +00:00
kjellander@webrtc.org
0372b93118 Partial revert of r7014 (Android APK refactor) 
This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 14:34:46 +00:00
aluebs@webrtc.org
bac072667b Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes 
The sizes saved in the aecdumps were always the input length, and this is not necessarily true when there is a change in sample rate. But the sample rates dumped are correct, so we can calculate the sizes from them knowing that we use 10ms chunks.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 13:39:01 +00:00
minyue@webrtc.org
adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate 
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 12:28:06 +00:00
stefan@webrtc.org
0a214ffa8a Setting marker bit on DTMF correctly 
BUG=1157
R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 11:46:54 +00:00
aluebs@webrtc.org
74cf916924 Fix issues in audioproc for float aecdumps 
* The right buffer size is used to dump to file when the output sample rate is different from the input one.
* The percentage of processed chunks is calculated correctly when float data available.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 11:05:01 +00:00
bjornv@webrtc.org
48f2568d89 audio_processing/nsx: Bug fix that could cause divide by zero 
In the fixed point version of the Noise Suppression. At one place we subtract a value in the wrong Q-domain, which later may cause a divide by zero. Going through the floating point code that particular variable should be zero if this happens, which is what the old code tried to accomplish, but in an awkward way.

The bug has been there since development, so the likelihood of actually get a divide by zero is very small.

BUG=chromium:407812
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 07:58:37 +00:00
andrew@webrtc.org
34a6764981 Remove the checks.h dependence on logging.h in a standalone build. 
logging.h apparently drags in a lot of undesirable dependencies. It was
only required for the trivial LogMessageVoidify; simply add an
identical FatalMessageVoidify instead.

Keep the include in a Chromium build to still have the override
mechanism use Chromium's macros.

Bonus: Add the missing DCHECK_GT (noticed by bercic).

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 19:00:45 +00:00