4673 Commits

Author SHA1 Message Date
henrike@webrtc.org
47658f1269 Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,
RTPStream, and NetEq as such.  Also mark all other virtual overrides in the same
files.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also deletes ACMTest.cc, which existed solely to define ~ACMTest(), which
was marked pure virtual in the header.  (Pure virtual destructors still need a
definition.)  Because there is another pure virtual method in this class, the
class is already abstract, so there's no benefit to making the desturctor pure.
Making it non-pure allows removing the separate source file.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:14:59 +00:00
glaznev@webrtc.org
3472dcd7b0 Fix frame rate selection for Android camera.
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.

BUG=2622
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 19:24:57 +00:00
brettw@chromium.org
0867f69cc6 Convert GN visibility to be lists.
This is a followup to my previous patch that missed this case.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7137 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 16:24:11 +00:00
andresp@webrtc.org
33aa095896 Simplify gyp rules on video_render_module.
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7135 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 14:48:48 +00:00
henrik.lundin@webrtc.org
23a5e3c3b0 Remove DestructEncoderInst and its codec-specific implementations.
This method is seemingly never called.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7131 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 08:52:26 +00:00
brettw@chromium.org
87ff9c8efa Fix up configs applying to GN build.
The audio_processing target didn't have the build configs applying to it which led to some logging errors.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/22339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7125 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 23:34:56 +00:00
fbarchard@google.com
a941970d4a Change explicit static cast from int to uint16_t to implicit cast of 0u.
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 21:37:27 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h.
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 10:40:56 +00:00
andrew@webrtc.org
641bda6f9c Initialize ChannelBuffer's memory to avoid uninitialized reads.
Removed the zero out memset in this change:
https://review.webrtc.org/24469004/

assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.

TBR=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/28429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 23:11:44 +00:00
brettw@chromium.org
519c9e207d Convert GN visibility to be a list.
GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 22:45:18 +00:00
andrew@webrtc.org
17454f79dc Add ctors to ChannelBuffer to enable copying on construction.
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.

R=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/24469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:27:04 +00:00
henrik.lundin@webrtc.org
c64246f42c Set a default speech type in iSAC wrapper
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.

BUG=crbug/411162
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:40:58 +00:00
henrik.lundin@webrtc.org
ed8bcd3ac5 Starting to implement the new ACM API
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:13:19 +00:00
bjornv@webrtc.org
37c39f3784 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.

Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:21:56 +00:00
bjornv@webrtc.org
0d394f3609 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:19:39 +00:00
sprang@webrtc.org
c30e9e2300 Ignore FEC packet in stats, if it is first packet on ssrc.
BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:20:18 +00:00
kjellander@webrtc.org
6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_"
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
f68cf93e1b Add video_capture_tests_apk_target
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.

BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:35:51 +00:00
henrik.lundin@webrtc.org
8f073c5054 Create a new interface for AudioCodingModule
This is a first draft of the interface, and is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:16:23 +00:00
henrik.lundin@webrtc.org
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
bjornv@webrtc.org
c2c4117477 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing
Affected components:
* AECMobile
  - Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 06:01:53 +00:00
aluebs@webrtc.org
021e76fd39 Add support for WAV output in audioproc
The default output is a WAV file, except if the --pcm_output flag is set.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 18:12:00 +00:00
brettw@chromium.org
afa77cd803 Add direct_dependent_config to desktop_capture in GN build.
This allows us to remove some configs in the Chrome build that should come
automatically from depending on this target.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7067 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:00:55 +00:00
andresp@webrtc.org
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
henrik.lundin@webrtc.org
f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo
The name of the copy method was confusing. This change makes the
code easier to read where the method is used.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 10:58:43 +00:00
kjellander@webrtc.org
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
kwiberg@webrtc.org
51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_
R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 08:42:44 +00:00
henrik.lundin@webrtc.org
7825b1abf9 Add support for multi-channel DTMF tone generation
This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 07:39:21 +00:00
fbarchard@google.com
9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 23:05:07 +00:00
henrike@webrtc.org
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
andresp@webrtc.org
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
kjellander@webrtc.org
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
aluebs@webrtc.org
bac072667b Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes
The sizes saved in the aecdumps were always the input length, and this is not necessarily true when there is a change in sample rate. But the sample rates dumped are correct, so we can calculate the sizes from them knowing that we use 10ms chunks.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7039 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 13:39:01 +00:00
minyue@webrtc.org
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
stefan@webrtc.org
0a214ffa8a Setting marker bit on DTMF correctly
BUG=1157
R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 11:46:54 +00:00
aluebs@webrtc.org
74cf916924 Fix issues in audioproc for float aecdumps
* The right buffer size is used to dump to file when the output sample rate is different from the input one.
* The percentage of processed chunks is calculated correctly when float data available.

BUG=webrtc:3359
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7036 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 11:05:01 +00:00
bjornv@webrtc.org
48f2568d89 audio_processing/nsx: Bug fix that could cause divide by zero
In the fixed point version of the Noise Suppression. At one place we subtract a value in the wrong Q-domain, which later may cause a divide by zero. Going through the floating point code that particular variable should be zero if this happens, which is what the old code tried to accomplish, but in an awkward way.

The bug has been there since development, so the likelihood of actually get a divide by zero is very small.

BUG=chromium:407812
R=ljubomir.papuga@gmail.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7035 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 07:58:37 +00:00
kjellander@webrtc.org
6f729e8a74 Disable video_engine_tests and webrtc_perf_tests on Android.
BUG=3770
TESTED=Running the tests locally on an Android device.
R=phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7026 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 15:13:55 +00:00
henrik.lundin@webrtc.org
ee0fb187a5 Divide-by-zero problem in NetEq's Normal::Process fixed
Adding a couple of tests that tries to trigger a certain divide-by-zero
issue. The tests triggered the issue, but this CL also includes a fix
for this.

BUG=3761
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7025 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 13:22:11 +00:00
kjellander@webrtc.org
324b72dda6 Disable video_capture_tests for Android.
BUG=3768
TESTED=Passing the steps in webrtc:3768
TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7023 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 12:37:50 +00:00
kjellander@webrtc.org
76a42577ad Remove build_with_chromium==1 conditions for Android
Most of these changes were done in r7014, but a few targets
were missed. This should make these tests run better
(but they might still be failing due to webrtc:3764).

BUG=webrtc:3741
TESTED=Local compilation using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7019 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 08:40:39 +00:00
aluebs@webrtc.org
841f58f64c Unpacking aecdumps generates wav files
BUG=webrtc:3359
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 07:51:51 +00:00
kjellander@webrtc.org
c3f42f37b5 Fix audio_decoder_unittests.isolate
In r6427 all .isolate files except
audio_decooder_unittests.isolate was updated to use the
<(DEPTH) variable instead of relative paths.
This started breaking the Android bots after committing
r7014.

BUG=3741
TBR=phoglund@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/23409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 15:06:14 +00:00
henrik.lundin@webrtc.org
8dbeb5b301 Adding more codecs to the AcmSenderBitExactness
New tests include iSAC-swb, PCM16b (8, 16, 32 kHz; mono and stereo),
PCM A/u (mono and stereo), iLBC, G.722 (mono and stereo), and Opus.

Also adding checks on number of output channels.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 14:19:00 +00:00
kjellander@webrtc.org
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
kjellander@webrtc.org
c4870bb221 GN: Audio device module
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 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

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=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/14259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 04:24:11 +00:00
kjellander@webrtc.org
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
andrew@webrtc.org
a22485eaf0 MIPS optimizations for AEC audio processing module
Added new optimizations for MIPS that were removed in r6797.
For more information about this see https://code.google.com/p/webrtc/source/detail?r=6797

R=andrew@webrtc.org, djordje.pesut@imgtec.com

Review URL: https://webrtc-codereview.appspot.com/15259004

Patch from Ljubomir Papuga <ljubomir.papuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7010 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:51:28 +00:00
andrew@webrtc.org
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
henrik.lundin@webrtc.org
f554d75288 Allow same src and dst in InputAudioFile::DuplicateInterleaved
This change allows the input and output to the static method
InputAudioFile::DuplicateInterleaved to be the same array. That is,
in-place manipulation is now possible. A unit test is also added.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7008 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 07:26:40 +00:00