1431e4dd1c
Revert 7675 "Make an AudioEncoder subclass for iSAC"
...
Above CL did not compile on Android. Followings are links to Android builds
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320
> Make an AudioEncoder subclass for iSAC
>
> BUG=3926
> R=henrik.lundin@webrtc.org , kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/25019004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 01:44:13 +00:00
05feff013e
Make an AudioEncoder subclass for iSAC
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-10 23:53:08 +00:00
6de75ca3ed
Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:29:24 +00:00
c78cf97ecb
Remove the useless dummy state parameter to WebRtcG711_*
...
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7609 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:23:36 +00:00
4765ca55f9
Roll chromium_revision: 28d1981..d3db2ff
...
Pick up the libvpx roll: https://codereview.chromium.org/674753002
Summary of changes (28d1981..d3db2ffR=andrew@webrtc.org
TBR=ajm@google.com
Review URL: https://webrtc-codereview.appspot.com/23229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7598 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-03 20:10:26 +00:00
78c222bfae
Update all .isolate files for the new format.
...
R=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27809004
Patch from Marc-Antoine Ruel <maruel@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 18:08:09 +00:00
508c91683c
Build fix for MIPS32R6.
...
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25989004
Patch from Ljubomir Papuga <lpapuga@mips.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:26:17 +00:00
decd9306ae
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket
...
Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.
BUG=3926
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 08:38:50 +00:00
663fdd02fd
Make an AudioEncoder subclass for Opus
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 07:28:36 +00:00
a296725d0e
audio_coding/codecs/isac/fix: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>"
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7544 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:05:43 +00:00
67ca26e087
common_audio: Removed trivial macro WEBRTC_SPL_UMUL_16_16
...
The macro made a cast to uint16_t before a plain multiplication. At the few places where it was used the variables were already uint16_t.
Affected components:
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7543 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 13:03:10 +00:00
0552356fda
isacfix: Refactor big-endian reading and writing
...
Make subroutines for encoding and decoding arrays of 16-bit big-endian
integers, and in the process fix a bug: When decoding an odd number of
bytes from be16, the least significant byte of the last int16 in the
array was properly taken to be zero instead of actually being read
(since it's outside the array). However, when encoding an odd number
of bytes, the least significant byte of the last int16 in the array
was written to the output as-is instead of being taken to be zero;
thus, we encoded one byte more than we should. This was probably not
harmful, and the value was dropped at decoding anyway; nevertheless,
writing a constant zero is the safe thing to do, and this patch does
so.
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7527 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 11:25:37 +00:00
def1e97ed2
Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests
...
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 12:48:29 +00:00
78ea06dd34
audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
Removed usage of trivial macro.
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-21 07:17:24 +00:00
2abebe7baf
audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
...
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-20 08:26:41 +00:00
264e66f7a5
Add encoded_timestamp to AudioEncoder base class
...
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 21:16:07 +00:00
9ea6f8a84d
New interface class AudioEncoder
...
This class will be the base for new C++ wrapper classes for all
encoders.
BUG=3926
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-16 11:26:24 +00:00
a3722b643d
iSAC tests: Type buffers as uint8_t[] to avoid casts
...
The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.
R=bjornv@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 13:29:04 +00:00
396a5e0001
WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
...
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:23:24 +00:00
3f7f899a15
WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
...
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 11:07:06 +00:00
1172988c79
Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
...
The affected functions are
WebRtcIsacfix_ReadFrameLen
WebRtcIsacfix_GetNewBitStream
WebRtcIsacfix_ReadBwIndex
and
WebRtcIsac_ReadFrameLen
WebRtcIsac_GetNewBitStream
WebRtcIsac_ReadBwIndex
WebRtcIsac_GetRedPayload
BUG=909
R=aluebs@webrtc.org , henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-13 10:53:42 +00:00
4bd2db9a55
Opus wrapper: Use const for inputs and uint8[] for byte streams
...
About half of the functions already followed the desired pattern; this
patch fixes the other half.
BUG=909
R=aluebs@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 11:21:10 +00:00
3ea35fdb1b
common_audio: Removed macro WEBRTC_SPL_LSHIFT_W16
...
The macro was a trivial << operation and where used has been replaced by <<. Affected components are
* ilbc
* isacfix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7404 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 08:47:02 +00:00
f71785cd3b
audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>
...
Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.
Affected codecs:
* ilbc
* isac/fix
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 15:36:30 +00:00
532ed43e85
Prevent reading outside iSAC bitstream, if the stream is corrupted.
...
BUG=chrome_373312(#24 )
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 00:21:02 +00:00
7c15510f38
common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32
...
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary.
Affects both fixed and float point versions of iSAC
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 09:40:38 +00:00
7ee24a7906
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
...
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , minyue@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7266
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 10:31:02 +00:00
a3c4d4dd2c
Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
...
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795
> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
>
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
>
> BUG=909
> R=andrew@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , minyue@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19229004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 01:32:57 +00:00
8c5740b485
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
...
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org , bjornv@webrtc.org , henrik.lundin@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 23:04:14 +00:00
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
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
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
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
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
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
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
3bd4156d75
Android APK tests built from a normal WebRTC checkout.
...
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@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 11:06:37 +00:00
524b8f7304
GN: Implement voice engine, common audio, audio coding and audio processing
...
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@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
af7fdfcde8
Add LTO support for Android Chromium.
...
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@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:41:13 +00:00
df9fef6638
common_audio: Removed macro WEBRTC_SPL_DIV
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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@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 12:57:32 +00:00
4f71e22bf9
Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV
...
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@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 10:25:10 +00:00
047a46f8b4
Remove Android.mk build files.
...
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@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
926707b167
Refactoring common_audio: Replace trivial multiplication macro
...
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@6964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 11:42:42 +00:00
d32c4389ac
Re-landing r6961
...
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@6963 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 11:19:05 +00:00
4a616be12b
Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..."
...
> 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@6962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 10:32:22 +00:00
4f01017e2d
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
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6961 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 10:23:22 +00:00
6e71d17bc9
Refactoring common_audio/signal_processing: Replaces trivial macros
...
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@6960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 07:44:52 +00:00
9b8102cf0e
Use a deterministic input in NetEqBgnTest
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This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.
The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)
BUG=3715
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 08:27:44 +00:00
52275341d8
Refactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16
...
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@6935 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 10:09:34 +00:00
742bac20b2
Remove __inline from WebRtcIsacfix_Log2Q8.
...
This function is used externally and needs to always be emitted, also
there's no point in explicitly marking this as inline.
R=tina.legrand@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/13279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6926 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 06:54:12 +00:00
