Commit Graph

7004 Commits

Author SHA1 Message Date
buildbot@webrtc.org
b4c7b09c13 (Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 12:11:58 +00:00
bjornv@webrtc.org
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
bjornv@webrtc.org
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
bjornv@webrtc.org
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
bjornv@webrtc.org
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
bjornv@webrtc.org
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
kwiberg@webrtc.org
584cd8da4b Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)
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@6959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 06:26:04 +00:00
buildbot@webrtc.org
3740d74106 (Auto)update libjingle 73927658-> 73927775
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22 22:27:04 +00:00
buildbot@webrtc.org
309a611670 (Auto)update libjingle 73891518-> 73927658
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22 22:24:54 +00:00
buildbot@webrtc.org
2b0554f0e7 (Auto)update libjingle 73794259-> 73891518
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22 14:08:15 +00:00
pbos@webrtc.org
97fdeb8329 Remove static initializer in WebRtcVideoEngine2.
Blocks import into chromium.

R=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/18249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6954 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-22 10:36:23 +00:00
kjellander@webrtc.org
374d39b7ae Increment sync_chromium.py version to force re-sync
This should make the remaining red Windows bots cycle green.
Currently, some of them are in a bad state for the Chromium
checkout.

BUG=webrtc:2863
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6953 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 19:51:06 +00:00
iannucci@chromium.org
161363808b Make the last_sync_chromium file a bit more comprehensive.
Adds a SCRIPT_VERSION and the target_os_list to the flag file content. The
script version is so that we can arbitrarially make all slaves/devs re-sync (in
case we change the implementation but don't want to roll chromium), and the
target_os_list is so that devs who change the target_os_list in their .gclient
file don't mysteriously fail to get the new deps.

R=kjellander@webrtc.org, agable@chromium.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/17189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6952 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 15:48:23 +00:00
niklas.enbom@webrtc.org
153c6162d2 Landing issue 15189004
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6951 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 14:49:28 +00:00
phoglund@webrtc.org
7bd5fefb17 Making sure muc members get recorded.
This is an upstream of a change I made; will describe in a separate
email thread.

Essentially, the members map wasn't getting populated in the callclient
example, so it was always empty. Now it will be populated correctly.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 09:53:28 +00:00
henrik.lundin@webrtc.org
038cee2401 Add send-side bit-exactness test for AudioCoding Module
This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:

test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest

The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.)  Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.

The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 08:59:14 +00:00
henrik.lundin@webrtc.org
9b8102cf0e Use a deterministic input in NetEqBgnTest
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
bjornv@webrtc.org
6b2659c660 Refactoring common_audio/signal_processing: Remove unused macro 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@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 06:13:57 +00:00
thakis@chromium.org
905f9efbae Fix clang -Wformat warnings.
An unsigned int was passed through %lu instead of %u (harmless on 32bit).
More seriously, a wide string was passed through %s, which means only the
first byte in the string got printed (since the 2nd byte is likely 0 in
UCS-2). Use %ls to include the whole string, even though it might not be
renderable in the target 8bit buffer.

BUG=chromium:82385
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6946 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 02:23:30 +00:00
thakis@chromium.org
add54ad770 Convert nsx_core_neon.S to unified syntax.
That way, it builds with both gcc and clang's integrated assembler.
No intentional behavior change.

BUG=chromium:124610
R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/15199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 02:23:26 +00:00
iannucci@chromium.org
286210d3ec Use --gclientfile instead of --spec, because windows is THE WORST.
--spec contains newlines, which are interpreted as actual newlines in the
command line, which causes gclient to fall apart at the seams.

TBR=agable@chromium.org, kjellander@webrtc.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/22429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6944 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-21 02:14:11 +00:00
iannucci@chromium.org
98d92d63e2 Make sync_chromium use the git-cache when on the bots.
This should help bootstrapping speed, as well as allow better clobbering
support.

R=agable@chromium.org
TBR=agable@chromium.org, kjellander@webrtc.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/17179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6943 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 23:53:59 +00:00
tnakamura@webrtc.org
8dcf61f946 Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome.
R=niklas.enbom@webrtc.org
TBR=niklas.embom

Review URL: https://webrtc-codereview.appspot.com/15219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6942 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 22:44:18 +00:00
kjellander@webrtc.org
3aa837ca28 Increase verbosity for gclient sync of Chromium
In r6939 the --verbose flag was passed to the problematic
(approx 2.2GB large) gclient sync of Chromium's src.git repo.
However the bots are still hitting killed sync jobs due to
lack of output. This is a speculative attempt to provoke
even more logging, in order to trigger buffer flushing for
the buildbot execution.

BUG=2863, chromium:339647
TEST=Ran gclient runhooks locally with CHROME_HEADLESS=1 set.
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 14:25:43 +00:00
kjellander@webrtc.org
bbca4dde0c Pass --verbose to gclient sync of Chromium
In r6938 the switch to using Chromium's Git repo was
deployed. However this fails on the bots since their timeout
for steps without output is 1200 seconds, which is not enough
to checkout the large Chromium Git repo.
Adding --verbose will print more output, thus getting a longer
timeout that should be enough for the runhooks step to complete.

BUG=2863, chromium:339647
TEST=None
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6939 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 13:04:49 +00:00
kjellander@webrtc.org
8925662318 Make WebRTC work with Chromium Git checkouts
WebRTC standalone shares a lot of dependencies and build
tools with Chromium. To make the build work, many of the
paths of a Chromium checkout is now emulated by creating
symlinks to files and directories.

All DEPS entries that previously used Var("chromium_trunk")
to reference a Chromium checkout or From("chromium_deps"..)
to reference the Chromium DEPS file are now removed and
replaced by symlink entries in setup_links.py.

The script also handles cleanup of the legacy
Subversion-based dependencies that's needed for the
transition.

Windows: One Windows-specific important change is that
gclient sync|runhooks must now be run from a shell
with Administrator privileges in order to be able to create
symlinks. This also means that Windows XP is no longer
supported.

To transition a previously created checkout:
Run "python setup_links.py --force" to cleanup the old
SVN-based dependencies that have been synced by gclient sync.
For Buildbots, the --force flag is automatically enabled for
their syncs.

BUG=2863, chromium:339647
TEST=Manual testing on Linux, Mac and Windows.
R=andrew@webrtc.org, iannucci@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 12:10:11 +00:00
henrik.lundin@webrtc.org
3fb2d0cd0e Add TSAN suppression for heap-use-after-free in libvpx
BUG=3671
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6937 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 11:07:29 +00:00
kjellander@webrtc.org
58c1c98d3b Remove DEPS reference to third_party/clang_format
Clang format has moved into Chromium's src/buildtools
and the last traces from third_party/clang_format were
removed in http://crrev.com/285030.

This removes it from the WebRTC checkouts as well (it is
now an tree of empty directories).

Our DEPS entry for removing the old binaries from pre-move
into src/buildtools was removed in
https://code.google.com/p/webrtc/source/detail?r=6788

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6936 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 10:47:47 +00:00
bjornv@webrtc.org
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
henrik.lundin@webrtc.org
6908b84179 Disable two tests in TurnPortTest
The tests are flaky.

BUG=3720
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6934 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 09:47:58 +00:00
buildbot@webrtc.org
95bbd18696 (Auto)update libjingle 73627179-> 73695227
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6933 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:49:30 +00:00
kwiberg@webrtc.org
877083c4d4 New utility class for easy debug dumping to WAV files
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@6932 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-20 07:42:46 +00:00
andrew@webrtc.org
71d9572e9c Minor bug fix and cosmetic changes in AEC MIPS optimizations.
Minor bug fix in WebRtcAec_FilterAdaptation_mips, which did not manifest with
gcc 4.7.2, but it did with version 4.9.0. While there, also made some cosmetic
changes to comply with Chromium coding style.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22399004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 15:42:50 +00:00
buildbot@webrtc.org
5a60aed80f (Auto)update libjingle 73626701-> 73627179
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6930 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 15:11:45 +00:00
buildbot@webrtc.org
84532e59dd (Auto)update libjingle 73626167-> 73626701
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6929 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 15:05:18 +00:00
henrike@webrtc.org
0481f15f02 (Auto)update libjingle 73399579-> 73626167
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 14:56:59 +00:00
houssainy@google.com
d5b292e450 Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc]
is now printed in the head-up display in Android appRTC.

This printing will be usefull in debugging switching ICE candidates.

R=andresp@webrtc.org, glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6927 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-19 11:43:32 +00:00
pbos@webrtc.org
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
henrike@webrtc.org
544f647a04 webrtc/base: removes accidental #error in r6909.
BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6924 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 20:55:58 +00:00
jiayl@webrtc.org
047abc93a2 Remove trailing null character from std::string
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 20:48:15 +00:00
andrew@webrtc.org
a1ad844229 Precompute the AEC FFT tables, rather than initializing at run-time.
These global arrays are shared amongst all AEC instances, and were at
serious risk of data races. A Chromium TSAN bot recently caught this.

Also move the function pointer selection for optimization to
create-time. (Ideally this would only be done once.)

BUG=chromium:404133,1503
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 19:02:51 +00:00
kjellander@webrtc.org
4a25199b76 GN: Fixes for Chromium builds.
When building WebRTC from a Chromium checkout (i.e. with
https://codereview.chromium.org/321313006/ applied) GN
cannot execute successfully.

This CL fixes:
- include path for video_processing module's SSE2 target.
- NSS/SSL targets

BUG=3441
TEST=
Passing WebRTC GN trybots.
Passing build from a Chromium checkout with https://codereview.chromium.org/321313006 applied and src/third_party/webrtc symlinked to the WebRTC checkout with this CL:
gn gen out/Default --args="clang_use_chrome_plugins=false" && ninja -C out/Default
gn gen out/Default --args="os=\"android\" cpu_arch=\"arm\"  clang_use_chrome_plugins=false" && ninja -C out/Default

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21179005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 17:56:28 +00:00
andrew@webrtc.org
d798095a37 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics.
The modification only uses the unique part of the spectrum (as is done for the C and asm code). It passes
byte to byte conformance test, and the single function performance
(if not specified, the code is compiled by GCC 4.6) on different
platforms:

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| C                          |      100% |      100% |       100% |
| Neon asm                   |       18% |       14% |        19% |
| Neon inline asm            |       31% |       25% |        27% |
| Neon intrinsic (GCC 4.6)   |       33% |       27% |        42% |
| Neon intrinscis (GCC 4.8)  |       17% |       14% |        19% |
| Neon intrinsics (LLVM 3.3) |       15% |       13% |        18% |

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13739004

Patch from Joe Yu <joe.yu@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6920 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 17:46:45 +00:00
andrew@webrtc.org
f86b262588 MIPS optimizations for ISAC (patch #3)
Implemented functions:
- WebRtcIsacfix_MatrixProduct1
- WebRtcIsacfix_MatrixProduct2

The optimizations are bit-exact to the C code.

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18019004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6919 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 17:32:19 +00:00
kjellander@webrtc.org
3ea9f578ce Remove unneeded WebKit dependency from DEPS.
This dependency was added in r4173 in order to support
the build/android/run_tests.py script.
In http://crrev.com/217588 that script was removed and
build/android/test_runner.py was the preferred script to
launch Android tests with.
Also in http://crrev.com/279515 the dependency on webkitpy
was removed and replaced by copies of the needed files in
build/android/pylib/utils/.

BUG=None
TEST=None
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6918 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 17:09:57 +00:00
minyue@webrtc.org
e9b493e763 Removing macro in acm_opus.cc
Remove it since macros are not recommended to use according to code style guide.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 12:06:31 +00:00
bjornv@webrtc.org
b5ab52d010 common_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and 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@6916 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 12:01:02 +00:00
braveyao@webrtc.org
8a2c84f59d Log the Android Audio API choice correctly.
BUG=3699
TEST=Manual Test
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 03:02:42 +00:00
kjellander@webrtc.org
d235eaef25 Suppress deprecation warnings in video_capture for iOS
The chromium_revision roll in r6913 broke the iOS build since the
videoMinFrameDuration and videoMaxFrameDuration properties
have been deprecated in iOS 7.0, which is now the default target
platform for iOS.

BUG=3705
TEST=Passing ios and ios_rel trybots.
TBR=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-16 20:47:16 +00:00
kjellander@webrtc.org
34a865a038 Roll chromium_revision 288251:289723
Mainly to pick up the libvpx.gyp change in r288724
to unblock https://webrtc-codereview.appspot.com/16229005/

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 288251:289723
which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

In a WebRTC checkout, that sums up to the following relevant changes:
* src/buildtools 59b932:567f0a
* testing/gtest 643:692
* testing/gmock 410:485
* third_party/boringssl/src 533cbe:c3d796
* third_party/libvpx 287125:289332
* third_party/libyuv 1035:1038
* third_party/nss 287121:289430
* third_party/opus/src 256783:289085
* tools/gyp 1959:1964

BUG=2863, chromium:339647
TEST=Local testing as trybots currently cannot handle DEPS changes properly due to http://crbug.com/385594
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6913 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-16 18:49:55 +00:00