external/webrtc.git
7 hours ago(Auto)update libjingle 73794259-> 73891518 master
buildbot@webrtc.org [Fri, 22 Aug 2014 14:08:15 +0000 (14:08 +0000)]
(Auto)update libjingle 73794259-> 73891518

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d

11 hours agoRemove static initializer in WebRtcVideoEngine2.
pbos@webrtc.org [Fri, 22 Aug 2014 10:36:23 +0000 (10:36 +0000)]
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

25 hours agoIncrement sync_chromium.py version to force re-sync
kjellander@webrtc.org [Thu, 21 Aug 2014 19:51:06 +0000 (19:51 +0000)]
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

29 hours agoMake the last_sync_chromium file a bit more comprehensive.
iannucci@chromium.org [Thu, 21 Aug 2014 15:48:23 +0000 (15:48 +0000)]
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

30 hours agoLanding issue 15189004
niklas.enbom@webrtc.org [Thu, 21 Aug 2014 14:49:28 +0000 (14:49 +0000)]
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

35 hours agoMaking sure muc members get recorded.
phoglund@webrtc.org [Thu, 21 Aug 2014 09:53:28 +0000 (09:53 +0000)]
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

36 hours agoAdd send-side bit-exactness test for AudioCoding Module
henrik.lundin@webrtc.org [Thu, 21 Aug 2014 08:59:14 +0000 (08:59 +0000)]
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

37 hours agoUse a deterministic input in NetEqBgnTest
henrik.lundin@webrtc.org [Thu, 21 Aug 2014 08:27:44 +0000 (08:27 +0000)]
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

39 hours agoRefactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_...
bjornv@webrtc.org [Thu, 21 Aug 2014 06:13:57 +0000 (06:13 +0000)]
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

43 hours agoFix clang -Wformat warnings.
thakis@chromium.org [Thu, 21 Aug 2014 02:23:30 +0000 (02:23 +0000)]
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

43 hours agoConvert nsx_core_neon.S to unified syntax.
thakis@chromium.org [Thu, 21 Aug 2014 02:23:26 +0000 (02:23 +0000)]
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

43 hours agoUse --gclientfile instead of --spec, because windows is THE WORST.
iannucci@chromium.org [Thu, 21 Aug 2014 02:14:11 +0000 (02:14 +0000)]
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

45 hours agoMake sync_chromium use the git-cache when on the bots.
iannucci@chromium.org [Wed, 20 Aug 2014 23:53:59 +0000 (23:53 +0000)]
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

46 hours agoBump WebRTC version number. Starting now, we will be setting WebRTC major version...
tnakamura@webrtc.org [Wed, 20 Aug 2014 22:44:18 +0000 (22:44 +0000)]
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

2 days agoIncrease verbosity for gclient sync of Chromium
kjellander@webrtc.org [Wed, 20 Aug 2014 14:25:43 +0000 (14:25 +0000)]
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

2 days agoPass --verbose to gclient sync of Chromium
kjellander@webrtc.org [Wed, 20 Aug 2014 13:04:49 +0000 (13:04 +0000)]
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

2 days agoMake WebRTC work with Chromium Git checkouts
kjellander@webrtc.org [Wed, 20 Aug 2014 12:10:11 +0000 (12:10 +0000)]
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

2 days agoAdd TSAN suppression for heap-use-after-free in libvpx
henrik.lundin@webrtc.org [Wed, 20 Aug 2014 11:07:29 +0000 (11:07 +0000)]
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

2 days agoRemove DEPS reference to third_party/clang_format
kjellander@webrtc.org [Wed, 20 Aug 2014 10:47:47 +0000 (10:47 +0000)]
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

2 days agoRefactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16
bjornv@webrtc.org [Wed, 20 Aug 2014 10:09:34 +0000 (10:09 +0000)]
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

2 days agoDisable two tests in TurnPortTest
henrik.lundin@webrtc.org [Wed, 20 Aug 2014 09:47:58 +0000 (09:47 +0000)]
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

2 days ago(Auto)update libjingle 73627179-> 73695227
buildbot@webrtc.org [Wed, 20 Aug 2014 07:49:30 +0000 (07:49 +0000)]
(Auto)update libjingle 73627179-> 73695227

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6933 4adac7df-926f-26a2-2b94-8c16560cd09d

2 days agoNew utility class for easy debug dumping to WAV files
kwiberg@webrtc.org [Wed, 20 Aug 2014 07:42:46 +0000 (07:42 +0000)]
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

3 days agoMinor bug fix and cosmetic changes in AEC MIPS optimizations.
andrew@webrtc.org [Tue, 19 Aug 2014 15:42:50 +0000 (15:42 +0000)]
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

3 days ago(Auto)update libjingle 73626701-> 73627179
buildbot@webrtc.org [Tue, 19 Aug 2014 15:11:45 +0000 (15:11 +0000)]
(Auto)update libjingle 73626701-> 73627179

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6930 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 73626167-> 73626701
buildbot@webrtc.org [Tue, 19 Aug 2014 15:05:18 +0000 (15:05 +0000)]
(Auto)update libjingle 73626167-> 73626701

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6929 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days ago(Auto)update libjingle 73399579-> 73626167
henrike@webrtc.org [Tue, 19 Aug 2014 14:56:59 +0000 (14:56 +0000)]
(Auto)update libjingle 73399579-> 73626167

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d

3 days agoActive connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc]
houssainy@google.com [Tue, 19 Aug 2014 11:43:32 +0000 (11:43 +0000)]
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

3 days agoRemove __inline from WebRtcIsacfix_Log2Q8.
pbos@webrtc.org [Tue, 19 Aug 2014 06:54:12 +0000 (06:54 +0000)]
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

4 days agowebrtc/base: removes accidental #error in r6909.
henrike@webrtc.org [Mon, 18 Aug 2014 20:55:58 +0000 (20:55 +0000)]
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

4 days agoRemove trailing null character from std::string
jiayl@webrtc.org [Mon, 18 Aug 2014 20:48:15 +0000 (20:48 +0000)]
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

4 days agoPrecompute the AEC FFT tables, rather than initializing at run-time.
andrew@webrtc.org [Mon, 18 Aug 2014 19:02:51 +0000 (19:02 +0000)]
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

4 days agoGN: Fixes for Chromium builds.
kjellander@webrtc.org [Mon, 18 Aug 2014 17:56:28 +0000 (17:56 +0000)]
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

4 days agoreplace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics.
andrew@webrtc.org [Mon, 18 Aug 2014 17:46:45 +0000 (17:46 +0000)]
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

4 days agoMIPS optimizations for ISAC (patch #3)
andrew@webrtc.org [Mon, 18 Aug 2014 17:32:19 +0000 (17:32 +0000)]
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

4 days agoRemove unneeded WebKit dependency from DEPS.
kjellander@webrtc.org [Mon, 18 Aug 2014 17:09:57 +0000 (17:09 +0000)]
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

4 days agoRemoving macro in acm_opus.cc
minyue@webrtc.org [Mon, 18 Aug 2014 12:06:31 +0000 (12:06 +0000)]
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

4 days agocommon_audio/signal_processing: Remove unused macros WEBRTC_SPL_GET_BYTE and WEBRTC_S...
bjornv@webrtc.org [Mon, 18 Aug 2014 12:01:02 +0000 (12:01 +0000)]
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

4 days agoLog the Android Audio API choice correctly.
braveyao@webrtc.org [Mon, 18 Aug 2014 03:02:42 +0000 (03:02 +0000)]
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

6 days agoSuppress deprecation warnings in video_capture for iOS
kjellander@webrtc.org [Sat, 16 Aug 2014 20:47:16 +0000 (20:47 +0000)]
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

6 days agoRoll chromium_revision 288251:289723
kjellander@webrtc.org [Sat, 16 Aug 2014 18:49:55 +0000 (18:49 +0000)]
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

6 days agoSet updated_rect for frames generated by WindowCapturer implementationsw
sergeyu@chromium.org [Fri, 15 Aug 2014 23:13:23 +0000 (23:13 +0000)]
Set updated_rect for frames generated by WindowCapturer implementationsw

Previous updated_rect wasn't set for frames generated by WindowCapturer
implementation. That makes them unustable with chromoting host that
uses update_rect. With that change the frames will always contain
updated_rect that coveras the whole frame.

Change by Ronak Vora <ronakvora@google.com>

R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/22079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6912 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days ago(Auto)update libjingle 73370064-> 73399579
buildbot@webrtc.org [Fri, 15 Aug 2014 18:26:12 +0000 (18:26 +0000)]
(Auto)update libjingle 73370064-> 73399579

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6911 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRename linuxwindowpicker to x11windowpicker & only use it with use_x11
henrike@webrtc.org [Fri, 15 Aug 2014 14:44:13 +0000 (14:44 +0000)]
Rename linuxwindowpicker to x11windowpicker & only use it with use_x11

These days we have Linux chromium builds that don't use X11. We don't
want webrtc to add an X11 dependency to those builds.

BUG=3625
R=henrike@webrtc.org, tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6909 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agoRevert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
tommi@webrtc.org [Fri, 15 Aug 2014 08:38:30 +0000 (08:38 +0000)]
Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."

The bot that had the problem was using an old version of STL, so relanding.

> Revert 6863 "Refactor StatsCollector and associated types."
>
> Breaks chrome compilation on Mac:
>
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
>           _Tp __x_copy = __x;
>               ^          ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
>           _M_insert_aux(end(), __x);
>           ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
>   reports.push_back(report1);
>           ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
>   StatsReport() : timestamp(0) {}
>
>
>
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> >
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> >
> > R=xians@webrtc.org
> >
> > Review URL: https://webrtc-codereview.appspot.com/15119004
>
> TBR=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21169004

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d

7 days agocommon_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16
bjornv@webrtc.org [Fri, 15 Aug 2014 05:17:20 +0000 (05:17 +0000)]
common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16

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@6907 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoSmall refactor on ViE to remove redudant conditions and long ifdefs.
andresp@webrtc.org [Thu, 14 Aug 2014 16:46:46 +0000 (16:46 +0000)]
Small refactor on ViE to remove redudant conditions and long ifdefs.

BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoExlcude two tests in VideoAdapter for WinDrMemoryFull.
marpan@webrtc.org [Thu, 14 Aug 2014 16:25:38 +0000 (16:25 +0000)]
Exlcude two tests in VideoAdapter for WinDrMemoryFull.

http://chromegw.corp.google.com/i/client.webrtc/builders/Win%20DrMemory%20Full/builds/793

BUG=3655
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6904 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoReturn an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
stefan@webrtc.org [Thu, 14 Aug 2014 15:10:49 +0000 (15:10 +0000)]
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().

Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.

Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.

R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdding a 5% as packet loss level for Opus
minyue@webrtc.org [Thu, 14 Aug 2014 12:16:12 +0000 (12:16 +0000)]
Adding a 5% as packet loss level for Opus

This is a follow up of
https://webrtc-codereview.appspot.com/16979004/

The purpose of this CL is to add 5% as a level for optimizing the packet loss rate to report to Opus. Adding such a level makes the grid finer.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6902 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoAdding online bitrate change to voe_cmd_test
minyue@webrtc.org [Thu, 14 Aug 2014 12:15:27 +0000 (12:15 +0000)]
Adding online bitrate change to voe_cmd_test

This is to verify a way of changing the bitrate on-the-fly under current WebRTC implementation.

TEST=changing bit rate for different codecs. sound quality changed when bit rate was set successful. catched error when bit rate is invalid for a running codec.

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

Review URL: https://webrtc-codereview.appspot.com/17059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6901 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoFix TimeToSendPadding return to be 0 if no padding bytes are sent.
andresp@webrtc.org [Thu, 14 Aug 2014 08:24:47 +0000 (08:24 +0000)]
Fix TimeToSendPadding return to be 0 if no padding bytes are sent.

BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agocommon_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32
bjornv@webrtc.org [Thu, 14 Aug 2014 07:26:28 +0000 (07:26 +0000)]
common_audio/signal_processing: Remove macro WEBRTC_SPL_SUB_SAT_W32

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@6899 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 73256845-> 73260148
buildbot@webrtc.org [Wed, 13 Aug 2014 23:57:23 +0000 (23:57 +0000)]
(Auto)update libjingle 73256845-> 73260148

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6898 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days agoRevert 6863 "Refactor StatsCollector and associated types."
niklas.enbom@webrtc.org [Wed, 13 Aug 2014 23:11:04 +0000 (23:11 +0000)]
Revert 6863 "Refactor StatsCollector and associated types."

Breaks chrome compilation on Mac:

/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
          _Tp __x_copy = __x;
              ^          ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
          _M_insert_aux(end(), __x);
          ^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
  reports.push_back(report1);
          ^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
  StatsReport() : timestamp(0) {}

> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
>
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
>
> R=xians@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15119004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d

8 days ago(Auto)update libjingle 73248599-> 73249894
buildbot@webrtc.org [Wed, 13 Aug 2014 21:55:18 +0000 (21:55 +0000)]
(Auto)update libjingle 73248599-> 73249894

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6896 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMake sure that muting muted streams succeeds.
pbos@webrtc.org [Wed, 13 Aug 2014 21:36:18 +0000 (21:36 +0000)]
Make sure that muting muted streams succeeds.

We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove TODO saying to remove WebRtcVideoFrame.
pbos@webrtc.org [Wed, 13 Aug 2014 21:17:22 +0000 (21:17 +0000)]
Remove TODO saying to remove WebRtcVideoFrame.

Comment was added prematurely, there's no decision to get rid of this
type.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemove files from talk/PRESUBMIT.py blacklist.
pbos@webrtc.org [Wed, 13 Aug 2014 20:38:53 +0000 (20:38 +0000)]
Remove files from talk/PRESUBMIT.py blacklist.

Many files can now be submitted here and do not have to be rolled in.

BUG=
R=henrike@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6893 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFixes failure triggered by include order re-ordering.
henrike@webrtc.org [Wed, 13 Aug 2014 18:39:43 +0000 (18:39 +0000)]
Fixes failure triggered by include order re-ordering.

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

Review URL: https://webrtc-codereview.appspot.com/18129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 73222930-> 73226398
buildbot@webrtc.org [Wed, 13 Aug 2014 17:26:08 +0000 (17:26 +0000)]
(Auto)update libjingle 73222930-> 73226398

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFurther DrMemory suppressions, likely from r6811
marpan@webrtc.org [Wed, 13 Aug 2014 17:17:40 +0000 (17:17 +0000)]
Further DrMemory suppressions, likely from r6811

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6890 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 73221069-> 73222930
buildbot@webrtc.org [Wed, 13 Aug 2014 16:47:12 +0000 (16:47 +0000)]
(Auto)update libjingle 73221069-> 73222930

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 73215194-> 73221069
buildbot@webrtc.org [Wed, 13 Aug 2014 16:22:04 +0000 (16:22 +0000)]
(Auto)update libjingle 73215194-> 73221069

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6888 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days ago(Auto)update libjingle 73072800 -> 73215194
henrike@webrtc.org [Wed, 13 Aug 2014 14:57:30 +0000 (14:57 +0000)]
(Auto)update libjingle 73072800 -> 73215194

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoDecreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4).
asapersson@webrtc.org [Wed, 13 Aug 2014 14:33:49 +0000 (14:33 +0000)]
Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4).
Increased kStandardRampUpDelayMs (30 to 40s).

BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6886 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFix the audio source failure due to unsupported constraints.
xians@webrtc.org [Wed, 13 Aug 2014 13:51:58 +0000 (13:51 +0000)]
Fix the audio source failure due to unsupported constraints.

Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.

This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.

BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback

R=henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemoving TODOs related to AcmReceiverBitExactness checksums
henrik.lundin@webrtc.org [Wed, 13 Aug 2014 13:02:00 +0000 (13:02 +0000)]
Removing TODOs related to AcmReceiverBitExactness checksums

Should have been part of r6883.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6884 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoUpdate checksums for AcmReceiverBitExactness on android
henrik.lundin@webrtc.org [Wed, 13 Aug 2014 10:38:15 +0000 (10:38 +0000)]
Update checksums for AcmReceiverBitExactness on android

This should have been a part of r6882.

BUG=3519
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6883 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoNetEq background noise generation off by default
henrik.lundin@webrtc.org [Wed, 13 Aug 2014 09:45:40 +0000 (09:45 +0000)]
NetEq background noise generation off by default

This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable.

With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise.

Reference files are updated where needed.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoFix STAP-A bug where we might overflow the packet buffer due to not accounting for...
stefan@webrtc.org [Wed, 13 Aug 2014 07:40:45 +0000 (07:40 +0000)]
Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.

BUG=3679
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoRemoving ASSERT for tcp candidate for port 0 and 9, as Android clients
mallinath@webrtc.org [Wed, 13 Aug 2014 06:05:55 +0000 (06:05 +0000)]
Removing ASSERT for tcp candidate for port 0 and 9, as Android clients
may not be called with set_allow_tcp_listen(false).

This CL will also sends tcp candidate in RFC 6544 format.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMove default-recv-channels to a separate class.
pbos@webrtc.org [Tue, 12 Aug 2014 23:17:13 +0000 (23:17 +0000)]
Move default-recv-channels to a separate class.

BUG=1788,3099
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d

9 days agoMake a int64 constant use ULL suffix so it wont get truncated.
fbarchard@google.com [Tue, 12 Aug 2014 22:39:06 +0000 (22:39 +0000)]
Make a int64 constant use ULL suffix so it wont get truncated.
BUG=3690
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoDrMemory suppresssions, likely from r6811.
marpan@webrtc.org [Tue, 12 Aug 2014 21:29:06 +0000 (21:29 +0000)]
DrMemory suppresssions, likely from r6811.

BUG=3655
R=henrike@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6877 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoFix GetStats() crash.
pbos@webrtc.org [Tue, 12 Aug 2014 20:55:10 +0000 (20:55 +0000)]
Fix GetStats() crash.

GetStats() can be called before codecs are set and the underlying
webrtc::VideoSendStream is created, leading to a null-pointer
dereference.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6876 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days ago.gitignore removed openssl
henrike@webrtc.org [Tue, 12 Aug 2014 16:04:00 +0000 (16:04 +0000)]
.gitignore removed openssl

BUG=N/A
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/19029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6875 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agotalk/third_party: removes the empty directory.
henrike@webrtc.org [Tue, 12 Aug 2014 15:57:02 +0000 (15:57 +0000)]
talk/third_party: removes the empty directory.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6874 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days ago(Auto)update libjingle 73072800-> 73072800
buildbot@webrtc.org [Tue, 12 Aug 2014 14:41:46 +0000 (14:41 +0000)]
(Auto)update libjingle 73072800-> 73072800

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6873 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoFixing uninitialized variable in file_audio_device.cc.
phoglund@webrtc.org [Tue, 12 Aug 2014 11:09:12 +0000 (11:09 +0000)]
Fixing uninitialized variable in file_audio_device.cc.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6872 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agocommon_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32
bjornv@webrtc.org [Tue, 12 Aug 2014 10:54:50 +0000 (10:54 +0000)]
common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32

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@6871 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoRemoves mismatching signs in signal_processing_unittests
bjornv@webrtc.org [Tue, 12 Aug 2014 10:27:21 +0000 (10:27 +0000)]
Removes mismatching signs in signal_processing_unittests

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@6870 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agoAdding SetOpusMaxBandwidth in VoE and ACM
minyue@webrtc.org [Tue, 12 Aug 2014 08:13:33 +0000 (08:13 +0000)]
Adding SetOpusMaxBandwidth in VoE and ACM

This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.

TEST = added a test in voe_cmd_test and listened to the result

BUG=
R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d

10 days agomodules/audio_processing: Updates output_data_fixed.pb test file
bjornv@webrtc.org [Tue, 12 Aug 2014 07:35:52 +0000 (07:35 +0000)]
modules/audio_processing: Updates output_data_fixed.pb test file

In r6591 a shift macro was removed affecting AECM. In addition to that change a bug was fixed. The fix added a few voice_counts in ApmTest.Process.

This CL updates the reference file, even though it is not used in practice since the test is currently turned off for Android (where AECM is used).

BUG=3672
TESTED=locally
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6868 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRemove more dependencies on openssl, add dependency on boringssl. Continues on r6798
henrike@webrtc.org [Mon, 11 Aug 2014 21:06:30 +0000 (21:06 +0000)]
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

11 days agomodules/audio_processing: Moves declaration of kDelayDiffOffsetSamples
bjornv@webrtc.org [Mon, 11 Aug 2014 15:39:00 +0000 (15:39 +0000)]
modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples

audio_processing did not compile when aec_untrusted_delay_for_testing=1 was set. The constant kDelayDiffOffsetSamples was declared only for Mac when WEBRTC_UNTRUSTED_DELAY was automatically turned on.

Moving the declaration outside the ifdef makes it build with the flag on for any platform.

BUG=3673
TESTED=locally and trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6866 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoMerge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics
henrik.lundin@webrtc.org [Mon, 11 Aug 2014 14:48:49 +0000 (14:48 +0000)]
Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics

The two tests both read and process the same (rather long) RTP input
file, and simply look at different outputs. This change merges the two
tests into one, in order to reduce testing time.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6865 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRebase webrtc/base with r6863 version of talk/base:
henrike@webrtc.org [Mon, 11 Aug 2014 14:32:13 +0000 (14:32 +0000)]
Rebase webrtc/base with r6863 version of talk/base:
cls integrated: r6809
svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff
patch -p0 -i 6809.diff

BUG=3379
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6864 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRefactor StatsCollector and associated types.
tommi@webrtc.org [Mon, 11 Aug 2014 14:08:33 +0000 (14:08 +0000)]
Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoUse test::Packet test::PacketSource classes in neteq_rtpplay
henrik.lundin@webrtc.org [Mon, 11 Aug 2014 12:29:38 +0000 (12:29 +0000)]
Use test::Packet test::PacketSource classes in neteq_rtpplay

This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoRevert 6860 "SSE2 version of SubbandCoherence()"
bjornv@webrtc.org [Mon, 11 Aug 2014 12:09:13 +0000 (12:09 +0000)]
Revert 6860 "SSE2 version of SubbandCoherence()"

> SSE2 version of SubbandCoherence()
>
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
>
> The output is bit exact.
>
> R=bjornv@webrtc.org, cd@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/18779004
>
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6861 4adac7df-926f-26a2-2b94-8c16560cd09d

11 days agoSSE2 version of SubbandCoherence()
bjornv@webrtc.org [Mon, 11 Aug 2014 10:38:31 +0000 (10:38 +0000)]
SSE2 version of SubbandCoherence()

The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
reported by audioproc is ~3.3%

The output is bit exact.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18779004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoFix a bug in parsing IceCandidate with IPV6 address.
jiayl@webrtc.org [Fri, 8 Aug 2014 23:09:15 +0000 (23:09 +0000)]
Fix a bug in parsing IceCandidate with IPV6 address.
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.

BUG=3669
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days ago(Auto)update libjingle 72931377-> 72931377
buildbot@webrtc.org [Fri, 8 Aug 2014 22:48:28 +0000 (22:48 +0000)]
(Auto)update libjingle 72931377-> 72931377

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6858 4adac7df-926f-26a2-2b94-8c16560cd09d

13 days agoEncoding and Decoding of TCP candidates as defined in RFC 6544.
mallinath@webrtc.org [Fri, 8 Aug 2014 22:29:20 +0000 (22:29 +0000)]
Encoding and Decoding of TCP candidates as defined in RFC 6544.

R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204

Review URL: https://webrtc-codereview.appspot.com/21479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoAllow root build dependencies to be overridden.
harryjin@google.com [Fri, 8 Aug 2014 00:08:58 +0000 (00:08 +0000)]
Allow root build dependencies to be overridden.

R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/22039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6856 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks ago(Auto)update libjingle 72847605-> 72850595
buildbot@webrtc.org [Thu, 7 Aug 2014 22:46:01 +0000 (22:46 +0000)]
(Auto)update libjingle 72847605-> 72850595

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks ago(Auto)update libjingle 72839629-> 72847605
buildbot@webrtc.org [Thu, 7 Aug 2014 22:09:08 +0000 (22:09 +0000)]
(Auto)update libjingle 72839629-> 72847605

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agowebrtc/base: removes linkage of crypto
henrike@webrtc.org [Thu, 7 Aug 2014 21:26:18 +0000 (21:26 +0000)]
webrtc/base: removes linkage of crypto

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d

2 weeks agoSupport for TURN/TLS.
tkchin@webrtc.org [Thu, 7 Aug 2014 20:39:08 +0000 (20:39 +0000)]
Support for TURN/TLS.

Wrap the socket in an SSL adapter, then simply call StartSSL() on the
SSLAdapter instance.

Cloned from: https://webrtc-codereview.appspot.com/21799004/

R=juberti@chromium.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/14059004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d