6ac22e6b47
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
...
R=andrew@webrtc.org , fbarchard@chromium.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 21:06:30 +00:00
0040a6ef97
This is a setup to solve
...
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).
BUG=
R=henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 14:41:57 +00:00
ceafa8cce9
MIPS optimizations for ISAC (patch #2 )
...
Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32
Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19749004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 16:43:13 +00:00
f563e85ab0
This is to re-open an earlier CL
...
https://webrtc-codereview.appspot.com/16619005/
which is reverted due to an issue in audio conference mixer.
This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/
BUG=webrtc:3155
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 21:11:27 +00:00
eec6ecdb1e
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
...
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org , pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:09:59 +00:00
8de8c9155e
Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow.
...
To save memory in iSAC-fix, decoder operated directly on the recieved bitstream. However, this breaks constantness of input when decoder performed in-place big to little Endian conversion. Furthermore, for bit-streams with odd lengths, this meant writing outside the memory. That is because the last byte will be shifted to the Most Significat Byte which might be outside the allocated memory.
If we care about memory, the solution is to do a big-to-little Endian conversion everytime we read a Word16 from the bitstream.
BUG=845,chrome:379458
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 15:47:09 +00:00
d42da54768
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
...
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
>
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/16619005
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 09:50:12 +00:00
8f8503d947
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
...
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:02:05 +00:00
721f970cba
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
...
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.
Related CL: https://webrtc-codereview.appspot.com/16669004
BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 10:30:14 +00:00
a1bfc50a72
Pass GYP DEPTH variable to isolate.
...
Similar change to https://codereview.chromium.org/322403003/
This will make it possible to handle different
directory levels for special builds of WebRTC, without
breaking GYP when the .isolate files are processed and
their contents is verified.
Also update all our .isolate files to use the <(DEPTH)
variable.
BUG=343106
TEST=Successful compile+test on Linux using:
ninja -C out/Release
tools/swarming_client/isolate.py run -s out/Release/tools_unittests.isolated
Also trybots passing all tests.
R=pbos@webrtc.org
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6427 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-13 09:02:15 +00:00
18026abd82
common_audio/signal_processing: Removes macro WEBRTC_SPL_RSHIFT_U16
...
This macro is only used at a few places and implies a cast to uint16_t before right shifting. All places already use uint16_t. Further, the amount of shifts applied in the macro has no sanity check for negativity, makes the macro dangerous to use.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 06:53:20 +00:00
782978cfcb
common_audio/signal_processing: Moves WEBRTC_SPL_UMUL_16_16_RSFT16 to iSAC fix
...
This macro is only used by the fixed point version of iSAC. Replacing the (five) locations in arith_routines_logist.c, where it is used, with the actual operation.
BUG=3348,3353
TESTED=trybots and manually
R=kwiberg@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 06:39:03 +00:00
919914d71b
MIPS optimizations for ISAC (patch #1 )
...
Implemented functions:
- WebRtcIsacfix_AutocorrMIPS
- WebRtcIsacfix_FilterArLoop
- WebRtcIsacfix_FilterMaLoopMIPS
- WebRtcIsacfix_AllpassFilter2FixDec16MIPS (only MIPS DSP)
- WebRtcIsacfix_PitchFilterCore (only MIPS DSPR2)
Gain achieved: from aprox. 15% (MIPS32) up to aprox. 40% (MIPS DSPR2)
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17559005
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6387 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-10 18:13:15 +00:00
aafd7a88c5
The correct fix of workaround in r6261.
...
The CL also includes same changes to filterbanks.c in iSAC fix and aecm_core_c.c
BUG=3370,3395,3439
TESTED=trybots
R=fdegans@chromium.org , glaznev@webrtc.org , kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6337 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:53:51 +00:00
edbe886a0b
common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND
...
This macro was only used at two places in fixed point iSAC, where it has been replaced with the operation.
BUG=3348,3353
TESTED=trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6336 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:42:53 +00:00
e6e139159f
Android: cleanup gtest_target_type conditions.
...
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library
Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
a28c697d93
- Get rid of 'using' from .h
...
- Add parenthesis to make order of evaluation clearer.
BUG=
R=minyue@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-02 15:22:33 +00:00
546961a9d3
Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix.
...
Arithmetic encoder does not right the last 2 or 3 bytes of |streamval| when terminating the bit-stream. Perhaps the last bytes makes no difference in decoding the stream. However, the decoder reads full |streamval| (int16_t) going out of boundary and reading uninitialized values. This avoids this problem. by inserting zero-bytes whenever decoder intends to read outside boundary.
BUG=1353,chrome373312,b/13468260
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6234 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 17:14:29 +00:00
a3b5673879
common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16
...
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.
BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 12:11:20 +00:00
1b21a57902
common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16
...
Macro was only mapping a function used in one place.
BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 06:40:31 +00:00
97e67cb476
Fix iOS assembly compile error.
...
In the roll of
https://webrtc-codereview.appspot.com/13369007
the fix in transform_neon.S was incorrectly removed
assuming it was only affecting Android when rolling to
265795. This CL fixes the iOS build when rolled to
266514.
Error looks like:
[893/2157] CC obj/webrtc/modules/audio_coding/codecs/isac/main/source/iSAC.entropy_coding.o
FAILED: /Volumes/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang -MMD -MF obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o.d -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DDISABLE_NACL -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DCLD_VERSION=1 -DENABLE_SPELLCHECK=1 -DDISABLE_FTP_SUPPORT=1 -DWEBRTC_RESTRICT_LOGGING -DWEBRTC_MODULE_UTILITY_VIDEO -DWEBRTC_ARCH_ARM -DWEBRTC_ARCH_ARM_V7 -DWEBRTC_ARCH_ARM_NEON -DWEBRTC_POSIX -DWEBRTC_MAC -DWEBRTC_IOS -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DNS_BLOCK_ASSERTIONS=1 -D_FORTIFY_SOURCE=2 -I../.. -I../.. -I../../webrtc -I../../webrtc/common_audio/resampler/include -I../../webrtc/common_audio/signal_processing/include -I../../webrtc/common_audio/vad/include -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk -Os -gdwarf-2 -fvisibility=hidden -Werror -Wnewline-eof -miphoneos-version-min=6.0 -arch armv7 -Wall -Wendif-labels -Wextra -Wno-unused-parameter -Wno-missing-field-initializers -Wheader-hygiene -Wno-c++11-narrowing -Wno-char-subscripts -Wno-unneeded-internal-declaration -Wno-covered-switch-default -Wstring-conversion -Wno-deprecated-register -Wno-absolute-value -Wno-selector-type-mismatch -std=c99 -fcolor-diagnostics -c ../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S -o obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:45:11: error: immediate expression for mov requires :lower16: or :upper16
mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
^
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:458:11: error: immediate expression for mov requires :lower16: or :upper16
mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
in
http://build.chromium.org/p/client.webrtc/builders/iOS%20Release/builds/911/steps/compile/logs/stdio
TBR=ajm
TEST=ios trybots passing tryjob based on r6010.
BUG=
Review URL: https://webrtc-codereview.appspot.com/12439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 10:25:30 +00:00
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
46509c8d58
adding FEC support to WebRTC Opus wrapper and tests.
...
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5656 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07 11:49:11 +00:00
04546884bf
This CL is to add Opus complexity knob and to test it.
...
As a by-product, a generic tool for testing and comparing the complexity of codecs is added, and new audio files are added to the resources.
Three complexity tests are included
1. Default Opus complexity
2. Opus complexity knob
3. Default iSAC complexity (to compare with Opus)
The complexity tests are only meant for development reasons
and not to be run at bots.
The .isolate file is only needed for the APK packaging and test execution on Android.
TEST=passes all trybots
BUG=
R=kjellander@webrtc.org , tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5655 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07 08:55:48 +00:00
e2fc13e42f
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
...
Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.
BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org , turajs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-14 23:12:34 +00:00
a80be4b23c
Add boundary checking to supress gcc 4.8.3 warning.
...
BUG=2888
Test=try, voe_cmd_test
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-11 16:38:45 +00:00
4ffd9c7423
Add full path to headers
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 09:01:39 +00:00
60730cfe3c
Remove the requirement to call set_sample_rate_hz and friends.
...
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org , bjornv@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-07 17:45:09 +00:00
917306d3fd
Change uses of the obsolete armv7 setting to arm_version==7.
...
BUG=http://crbug.com/234135
R=andrew@webrtc.org , fischman@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5369004
Patch from Mostyn Bramley-Moore <mostynb@opera.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 09:26:07 +00:00
57eb858698
Remove ".." from include_dirs in build/common.
...
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2332004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-11 10:20:27 +00:00
31628aae7e
Upgrade scoped_ptr to Chromium's latest version.
...
Analogous to the recent libjingle change: http://cl/54929753-p10 .
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 12:50:00 +00:00
621df678c8
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
...
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 10:27:23 +00:00
137b3793d9
Only use -lm on Linux in ISAC.
...
Remove unneeded WEBRTC_LINUX define.
BUG=crbug.com/298656
TESTED=Passed trybots.
R=wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2313004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4865 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 18:16:28 +00:00
3e7703640f
Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
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R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4844 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 22:05:05 +00:00
76fe9309b9
Use link_settings instead of all_dependent_settings to pacify xcode gyp generator
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Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2261004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 21:11:08 +00:00
ccddd0a941
Roll webrtc's chromium_revision 217707:224141
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Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2177007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 20:27:32 +00:00
31b4a5ac82
Recognize armv7 target_arch for ios support in webrtc common.gyp
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BUG=2343
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2176004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4684 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:46:36 +00:00
ecbe0aa543
Added Opus stereo support
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TESTED=git try
BUG=webrtc:1360
R=tina.legrand@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1868004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4521 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 06:48:09 +00:00
2ab209ef14
Remove include_dirs from test/test.gyp.
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This is a cleanup step for having root-relative includes, include_dirs shouldn't be needed anymore.
BUG=1662
R=phoglund@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1984004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4512 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:49:48 +00:00
a3b7406219
Remove unused unreferenced code in webrtc/
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The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.
BUG=
R=andrew@webrtc.org , henrike@webrtc.org , phoglund@webrtc.org , stefan@webrtc.org , turaj@webrtc.org , wu@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1945004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:47:51 +00:00
bd21fb5f8d
Adding call to Opus PLC
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NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.
BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1727004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 11:01:07 +00:00
45426eadf5
In call to Opus decoder: frame length too large
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BUG=https://code.google.com/p/webrtc/issues/detail?id=1201
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1752004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4292 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03 13:32:04 +00:00
31c5f1c91a
Remove ancient and unused CNG test.
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R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1585005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4154 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-03 16:07:07 +00:00
9aca5b34e1
Remove #pragma once
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BUG=1830
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1568004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 13:19:09 +00:00
4ce838934c
Address sanitizer out of bounds read in iSAC
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BUG=issue1770
TBR=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/1472006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 17:42:22 +00:00
3004c79c6a
Fix clang errors in non-GYP_DEFINES=clang=1 build
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BUG=1623
R=stefan@webrtc.org , tina.legrand@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1368004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-07 12:36:21 +00:00
342353780d
Consolidate common_audio into a single target.
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In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org , kma@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-30 23:43:26 +00:00
db11fab49e
Adding Opus unit test
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This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).
BUG=
Review URL: https://webrtc-codereview.appspot.com/1222006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-17 10:39:41 +00:00
6e788df19e
Remove vim/emacs modelines from .gypi files
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BUG=1655
Review URL: https://webrtc-codereview.appspot.com/1326005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 12:40:34 +00:00
e4b6064f8e
Replace legacy G_CONST with const.
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BUG=1608
Review URL: https://webrtc-codereview.appspot.com/1310005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3814 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-10 18:06:57 +00:00
