summaryrefslogtreecommitdiff
path: root/common_audio
AgeCommit message (Collapse)Author
2014-09-08Update makefiles after merge of Chromium at 041843cbf814Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ib50e00ed5e94f0daf8eefe82b049a93bc3f416ef
2014-09-08Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275 This commit was generated by merge_from_chromium.py. Change-Id: Icd984259a9896fb874700b1e7a2e42bbabfb204b
2014-09-07GN: Prefix WebRTC specific variables with "rtc_"kjellander@webrtc.org
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/webrtc@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04Fix MSVC warnings about value truncations, webrtc/common_audio/ edition.bjornv@webrtc.org
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/webrtc@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03Partial revert of r7014 (Android APK refactor)kjellander@webrtc.org
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/webrtc@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c This commit was generated by merge_from_chromium.py. Change-Id: If6328cd0f1a485164c204accf41b23892a162235
2014-09-02Unpacking aecdumps generates wav filesaluebs@webrtc.org
BUG=webrtc:3359 R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14239004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01Android APK tests built from a normal WebRTC checkout.kjellander@webrtc.org
Restructure how the Android APK tests are compiled now that we have a Chromium checkout available (since r6938). This removes the need of several hacks that were needed when building these targets from inside a Chromium checkout. By creating a symlink to Chromium's base we can compile the required targets. This also removes the need of the previously precompiled binaries we keep in /deps/tools/android at Google code. All the user needs to do is to add the target_os = ["android"] entry to his .gclient as described at https://code.google.com/p/chromium/wiki/AndroidBuildInstructions Before committing this CL, the Android APK buildbots will need to be updated. This also solves http://crbug.com/402594 since the apply_svn_patch.py usage will be similar to the other standalone bots. It also solves http://crbug.com/399297 BUG=chromium:399297, chromium:402594 TESTED=Locally compiled all APK targets by running: GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks ninja -C out/Release checkdeps R=henrike@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31GN: Implement voice engine, common audio, audio coding and audio processingkjellander@webrtc.org
NOTICE: Assembly offsets generation for audio processing will not be ported to GN and the process of removing them is tracked in https://code.google.com/p/webrtc/issues/detail?id=3580. The GN files are based upon the GYP files as of r7009. BUG=3441 TESTED=Passing builds with: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now: gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default Compilation of Chromium's 'all' target with src/third_party/webrtc symlinked to the WebRTC checkout with this CL applied, both with the default GN settings and using --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" R=andrew@webrtc.org, brettw@chromium.org Review URL: https://webrtc-codereview.appspot.com/15999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29Add LTO support for Android Chromium.andrew@webrtc.org
This is to add support for a Link-Time Optimizations experiment in Android Chromium. As it is disabled by default, it won't change anything for most configurations. BUG=chromium:407544 R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28Add CHECK and friends from Chromium.andrew@webrtc.org
Replace FATAL_ERROR_IF with the more familiar (to Chromium developers) CHECK and DCHECK. The full Chromium implementation is fairly elaborate but I copied enough to get us most of the benefits. I believe the main missing component is a more advanced stack dump. For this bit I relied on the V8 implementation. There are a few minor modifications from the Chromium original: - The FatalMessage class is specialized for logging fatal error messages and aborting. Chromium uses the general LogMessage class, which we could consider moving towards in the future. - NOTIMPLEMENTED() and NOTREACHED() have been removed, partly because I don't want to rely on our logging.h until base/ and system_wrappers/ are consolidated. - FATAL() replaces LOG(FATAL). Minor modifications from V8's stack dump: - If parsing of a stack trace symbol fails, just print the unparsed symbol. (I noticed this happened on Mac.) - Use __GLIBCXX__ and __UCLIBC__. This is from examining the backtrace use in Chromium. UNREACHABLE() has been removed because its behavior is different than Chromium's NOTREACHED(), which is bound to cause confusion. The few uses were replaced with FATAL(), matching the previous behavior. Add a NO_RETURN macro, allowing us to remove unreachable return statements following a CHECK/FATAL. TESTED=the addition of dummy CHECK, DCHECK, CHECK_EQ and FATAL did the did the right things. Stack traces work on Mac, but I don't get symbols on Linux. R=henrik.lundin@webrtc.org, kwiberg@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7003 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28common_audio: Removed macro WEBRTC_SPL_DIVbjornv@webrtc.org
The macro has no built-in divide by zero check. The only thing that is done is casting to int32_t. In addition a bug was discovered where it was supposed to do a division with rounding, but instead did a division with truncation + addition by 2. This is corrected in this CL. BUG=3348,3353 TESTED=locally on Linux R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27Update makefiles after merge of Chromium at b241671f0248Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I1c4f94279ad561b4f0f5d73edca8ae1e9e86e36c
2014-08-26Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIVbjornv@webrtc.org
This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use. This CL replaces the macro with '/' at place. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() ↵bjornv@webrtc.org
optimizations on armv7 According to the issue, common_audio_unittests failed on armv7. It currently pass, so we should turn it on again. There is no print out in the issue, so the cause of failure is unknown. BUG=740 TESTED=locally on N7 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26Remove Android.mk build files.pbos@webrtc.org
These files are generally not maintained and break, some contain files that don't exist anymore and do not build anymore. If we need to add some of these back we should really set up a bot for them. R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/15249004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25Refactoring common_audio: Replace trivial multiplication macrobjornv@webrtc.org
This multiplication macro literally use the '*' operator, so there is no need for it. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25Re-landing r6961bjornv@webrtc.org
common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. BUG=3348,3353 TESTED=locally on linux TBR=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..."bjornv@webrtc.org
> common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 > > This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. > > BUG=3348,3353 > TESTED=locally on linux and trybots > R=kwiberg@webrtc.org, tina.legrand@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/16359004 TBR=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8bjornv@webrtc.org
This macro is nothing but memcpy() and further used at one single place in webrtc, so it makes no sense to keep it. Replaced the operation where it is used. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25Refactoring common_audio/signal_processing: Replaces trivial macrosbjornv@webrtc.org
The macros WEBRTC_SPL_ADD_SAT_W16 and WEBRTC_SPL_ADD_SAT_W32 make direct use of the corresponding functions WebRtcSpl_AddSatW16() and WebRtcSpl_AddSatW32(). This CL replaces these macros in the code. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)kwiberg@webrtc.org
And in the process, make it dump WAV files instead of raw PCM. R=andrew@webrtc.org, bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21Update makefiles after merge of Chromium at 291084Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I76d1c82b1481ca20fc3520fb631c5cba99805342
2014-08-21Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 67afd1fc176021f625e064f20ae747e23d87d727 This commit was generated by merge_from_chromium.py. Change-Id: I6c68c68562042b290acef201bb3d998bee1dc9be
2014-08-21Refactoring common_audio/signal_processing: Remove unused macro ↵bjornv@webrtc.org
WEBRTC_SPL_MUL_32_32_RSFT32BI The WEBRTC_SPL_MUL_32_32_RSFT32BI macro was removed in r6169, since it was unused. This CL removes the arm and mips optimizations of it. BUG=3348, 3353 TESTED=locally and trybots TBR=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20Refactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16bjornv@webrtc.org
Yet another macro that utilizes a function directly. BUG=3348,3353 TESTED=locally on linux and trybots R=kwiberg@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18159004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6935 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20New utility class for easy debug dumping to WAV fileskwiberg@webrtc.org
There are currently a number of places in the code where we dump audio data in various stages of processing for debug purposes. Currently these all write raw, uncompressed PCM files, which isn't supported by the most common audio players, and requires the user to supply metadata such as sample rate, sample size and endianness, etc. This patch adds a simple class that makes it easy to write WAV files instead. WAV files still contain the same uncompressed PCM data, but they have a small header that contains all the requisite metadata, and are supported by virtually all audio players. Since some of the debug code that will be writing WAV files is written in plain C, a C API is included as well. R=andrew@webrtc.org, bjornv@webrtc.org, henrike@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19Merge third_party/webrtc from ↵Torne (Richard Coles)
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414 This commit was generated by merge_from_chromium.py. Change-Id: Iffb1ebee1a492d44a944cc8571605a2b40061d3e
2014-08-18common_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and ↵bjornv@webrtc.org
WEBRTC_SPL_SET_BYTE These two macros are not used anywhere in webrtc. Previously used in old neteq (I think). BUG=3348,3353 TESTED=manually on linux and trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-15common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16bjornv@webrtc.org
Macros should in general be avoided. WEBRTC_SPL_UMUL_32_16_RSFT16 is only used in iSAC fixed point as part of multiplying with LSB and MSB. A better approach is to have one function for that complete operation in iSAC. This CL removes the macro and replace the operation locally. BUG=3148, 3353 TESTED=locally on Linux and trybots R=tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb This commit was generated by merge_from_chromium.py. Change-Id: If1f6b3b63673dba2f4d77d2ddea983f77d0eaab9
2014-08-14common_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32bjornv@webrtc.org
This macro is literally using the function WebRtcSpl_SubSatW32(), hence there is no need for a macro. BUG=3348, 3353 TESTED=locally on Linux and trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12Update makefiles after merge of Chromium at 288938Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ic0141d5da4e1ab26e23ffa3c860ea955981acdb0
2014-08-12common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32bjornv@webrtc.org
The macro is only used at four places in iSAC fixed point and the macro have been replaced at those places. In addition, it is used in a unit test, but throws a warning treated as error (issue3674). The macro has both MIPS and armv7 optimizations. Removing them impacts only MIPS platforms without DSP ASE. This may cause a very small increase in complexity when using iSAC fix. The armv7 optimizations are not used anywhere, since specific ones are used inline in iSAC fix. BUG=3348,3353,3674 TESTED=locally and trybots R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16299004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12Removes mismatching signs in signal_processing_unittestsbjornv@webrtc.org
Negative inputs was used in WebRtcSpl_NormU32() causing warnings. BUG=3674 TESTED=locally and trybots R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11Remove more dependencies on openssl, add dependency on boringssl. Continues ↵henrike@webrtc.org
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/webrtc@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04Update makefiles after merge of Chromium at 287308Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I85bf9681e9c3bbef9f67d93b3d275289e6911e3c
2014-07-29Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 This commit was generated by merge_from_chromium.py. Change-Id: Iada7abd78f123301a98db982a6272cd9487de72f
2014-07-16int16<->float conversions: Use size_t for array length argument, not intkwiberg@webrtc.org
size_t is more appropriate for array lengths, since int might theoretically be too small for a really large array. But more importantly, if the caller's value is naturally of type size_t and the function requires an int, VC++ will trigger warning C4267 (http://msdn.microsoft.com/en-us/library/6kck0s93.aspx) because the implicit cast might be lossy, forcing the caller to do a manual cast. Typing the function with size_t in the first place resolves the problem. R=aluebs@webrtc.org, andrew@webrtc.org, minyue@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6702 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10Update makefiles after merge of Chromium at 282385Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I85d3e5fb3d9291809471c199df114e462a4739d6
2014-07-09Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36 This commit was generated by merge_from_chromium.py. Change-Id: Iffa5413ebfb78de36b84b4e85d94adc093f912df
2014-07-03common_audio: Removes macro WEBRTC_SPL_SHIFT_W16bjornv@webrtc.org
We should avoid macros in general (see style guide). This shift macro is not a severe one, since there is a check for negativity. BUG=3348,3353 TESTED=trybots and manually R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6591 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-25Update makefiles after merge of Chromium at 279716Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: Ie631c1eeed85862b7bade8e178791c7b230573b3
2014-06-25Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516 This commit was generated by merge_from_chromium.py. Change-Id: Ibdea97e7e6e800b2b7d7d3122a1d77e467cfbde4
2014-06-23GN: Add BUILD.gn files + kjellander to OWNERSkjellander@webrtc.org
This should work as a foundation for all the work that is left to do to make the parts of WebRTC that Chromium uses to build with GN. I implemented some the smaller modules myself in this CL. The remaining work (TODO's in the .gn files) will be distributed to various team members. I'm adding myself to OWNERS files for BUILD.gn files in all the directories where I'm adding a BUILD.gn file. BUG=3441 TEST= Successful compilation of WebRTC as standalone: gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default I built successfully from a Chromium checkout (with https://codereview.chromium.org/321313006/ applied) using: gn gen out/Default && ninja -C out/Default webrtc R=brettw@chromium.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13749004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19Update makefiles after merge of Chromium at 278252Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I2e09453759ef2a9b23eb8b2cf0d92f70acc3ea89
2014-06-16Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af This commit was generated by merge_from_chromium.py. Change-Id: Id1e94a534a8e364431bcb714b54729e7a410664d
2014-06-16Update makefiles after merge of Chromium at 277428Android Chromium Automerger
This commit was generated by merge_from_chromium.py. Change-Id: I35c59b36614d836accbb543178393a6c061586f1
2014-06-16Merge third_party/webrtc from ↵Android Chromium Automerger
https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a This commit was generated by merge_from_chromium.py. Change-Id: I58be5a5957c0a6b1be9beac86538af8d38058e9e
2014-06-16common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16bjornv@webrtc.org
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/webrtc@6444 4adac7df-926f-26a2-2b94-8c16560cd09d