| 2622ea73e33bf4269dcccff89a7ba224a80975b9 |
24-Feb-2017 |
Chih-Hung Hsieh <chh@google.com> |
Leave only an empty top level OWNERS file.
We should not copy OWNERS files from upstream,
or the owners should be registered in Gerrit Code Review.
Bug: 33166666
Test: default build targets
Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
WNERS
|
| 4e188dd3746eaf342d32e5f666de0e63480f7412 |
12-Feb-2016 |
Chih-Hung Hsieh <chh@google.com> |
Suppress unused-parameter warnings.
Suppress warnings until upstream can fix them.
BUG: 27074506
Change-Id: If7e6f190100fba025d25d2634d1c9a657cc24854
ndroid.mk
|
| 675d4373f87b2468a334f2ed48bfa4e6946d80f1 |
05-Nov-2015 |
Chih-Hung Hsieh <chh@google.com> |
WIP: Changes after merge commit 'cb3f9bd'
Changes after "git merge cb3f9bd"
* git mv old Android.mk from src/ to webrtc/
* Remove old unused files in src/*.
* Modify webrtc/.gitignore to keep *.mk files.
* Copy old files from master, lost in auto-merge.
src/modules/audio_processing/test/unit_test.cc
src/modules/audio_coding/codecs/isac/fix/test/{Android.mk,kenny.c}
to webrtc, but most of the old test code do not compile with new
webrtc API and are commented out.
* Move src/modules/audio_processing/test/android/apmtest/jni/*.mk to
webrtc/... but the Android.mk files does not work.
Commented out its build target.
* Changes to Android.mk files:
* Change references of src/ to webrtc/.
* Fix include path
* Fix source file list, remove old non-existing files,
add new source files to resolve link errors.
* Add new Android.mk files to build some new static libraries
to link into current Android webrtc .so files.
* Remove unnecessary LOCAL_SHARED_LIBRARIES in Android.mk files
that build static libraries.
* Remove old unnecessary clang workarounds like
-Wno-tautological-pointer-compare
-no-integrated-as
* Fix include path of debug.pb.h in some source files.
* Add -DWEBRTC_POSIX in android-webrtc.mk
* Manually merge Android specific changes in
src/typedefs.h to webrtc/typedefs.h
* Fix trivial syntax error in scoped_ptr.h, calling static_assert.
* Use -std=c++0x in webrtc/system_wrappers/source/Android.mk
* #undef getchaar in spreadsort.hpp
* Verified and not to carry old Android hacks from src/... to webrtc/...
src/system_wrappers/source/android/cpu-features.c
src/modules/interface/module.h
src/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
src/system_wrappers/source/trace_posix.cc
src/typedefs.h
More pathes from Alex Luebs:
* Use new unit test kenny.cc.
Delete old kenny.cc.
Comment out unessential code in kenny.cc to fix link error for now.
* Replace old unit test files with new ones in
webrtc/modules/audio_processing/Android.mk.
Delete old audio_processing/test/unit_test.cc.
* Fix compilation errors in
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
Change-Id: I7bbf776eeb9dcfa21a82dd1f2dec378235cbbc3e
ndroid.mk
|
| 74640895fafbb90a6630a6a91b80da0a7cff229c |
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417173004 .
Cr-Commit-Position: refs/heads/master@{#10444}
udio_decoder_isac.cc
udio_encoder_isac.cc
udio_encoder_isac_unittest.cc
sac.c
sac_float_type.h
sac_unittest.cc
tructs.h
|
| 5460f9b81dc9bd232fa856347729f120159ae168 |
20-Oct-2015 |
msniegocki <msniegocki@opera.com> |
Workaround for false positive -Wmaybe-uninitialized being triggered on some compilers
Some toolchains (in this case referring to a g++ 4.9, or "arm-linux-
androideabi-g++ (GCC) 4.9 20140827 (prerelease)" according to my
--version, from the Android NDK r10e-rc4 and potentially with custom
patches; others may be affected as well) fail to prove that myVec in
WebRtcIsac_CorrelateInterVec is never used uninitialized. This is likely
due to the compiler thinking the assignment in line 468 might not
happen. Changing the loop condition in line 466 to rowCntr <
SOME_CONSTANT also helps, suggesting that the compiler can't infer that
there are only 2 values interVecDim can have at that point, and neither
of them are 0. Of course, this is not an acceptable fix, as it changes
behaviour.
This seems to be a compiler bug, or at least an issue with its
heuristics. However, we can't really change toolchains at the moment,
and ultimately this change improves support for certain older compilers.
BUG=
Review URL: https://codereview.webrtc.org/1406423004
Cr-Commit-Position: refs/heads/master@{#10337}
ncode_lpc_swb.c
|
| bd7de0c6eff510f9ec95b13a16862ca55871c47d |
14-Oct-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Delete full-band mode from the iSAC codec
This mode is no longer used.
BUG=4210
Review URL: https://codereview.webrtc.org/1392173004
Cr-Commit-Position: refs/heads/master@{#10275}
sac.c
ettings.h
tructs.h
|
| 740436899825b96e12469c417c0ea82fd8a22edf |
22-Sep-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Move AudioDecoderIsac* to its own files
Currently, it's sitting in AudioEncoderIsac*'s files, which is less
than obvious. This CL puts the encoder and decoder in separate files
together with the C implementation; CLs are afoot to make it so for
the other built-in codecs as well.
BUG=webrtc:4557
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1339253003 .
Cr-Commit-Position: refs/heads/master@{#10018}
udio_decoder_isac.cc
udio_encoder_isac.cc
sac_float_type.h
|
| 12cfc9b4dacd6942377df1f29a64bdbec591920e |
08-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Fold AudioEncoderMutable into AudioEncoder
It makes more sense to combine the two interfaces, since there wasn't
a clear line separating them. The result is a combined interface with
just over a dozen methods, half of which need to be implemented by
every subclass, while the other half have sensible (and trivial)
default implementations and are implemented only by the few subclasses
that need non-default behavior.
Review URL: https://codereview.webrtc.org/1322973004
Cr-Commit-Position: refs/heads/master@{#9894}
udio_encoder_isac.cc
|
| 4376648df021fd82f25a38694e33678f802d06ea |
27-Aug-2015 |
Karl Wiberg <kwiberg@google.com> |
AudioDecoder: Replace Init() with Reset()
The Init() method was previously used to initialize and reset
decoders, and returned an error code. The new Reset() method is used
for reset only; the constructor is now responsible for fully
initializing the AudioDecoder.
Reset() doesn't return an error code; it turned out that none of the
functions it ended up calling could actually fail, so this CL removes
their error return codes as well.
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1319683002 .
Cr-Commit-Position: refs/heads/master@{#9798}
sac.c
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
andwidth_estimator.c
andwidth_estimator.h
odec.h
ecode_bwe.c
ilter_functions.c
sac.c
sac_unittest.cc
pc_analysis.c
pc_analysis.h
itch_estimator.h
|
| 608c3cfe77c165965ea04fcd0a2a71aad05a1d16 |
24-Aug-2015 |
kwiberg <kwiberg@webrtc.org> |
iSAC: Make separate AudioEncoder and AudioDecoder objects
The only shared state is now the bandwidth estimation info.
This reduces the amount and complexity of the locking
substantially.
Review URL: https://codereview.webrtc.org/1208993010
Cr-Commit-Position: refs/heads/master@{#9762}
udio_encoder_isac.cc
udio_encoder_isac_unittest.cc
|
| 805d8fb6eba5dfe4a1d4a875769d2f355683e21d |
14-Aug-2015 |
pkasting <pkasting@chromium.org> |
Remove WebRtcIsac_Highpass_float().
This function is unreferenced and not even declared in a header file.
Split from https://codereview.webrtc.org/1228793004/ .
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1296513002
Cr-Commit-Position: refs/heads/master@{#9716}
ilter_functions.c
|
| b297c5a01f88219da26cffe433804963d1b70f0f |
23-Jul-2015 |
pkasting <pkasting@chromium.org> |
Miscellaneous changes split from https://codereview.webrtc.org/1230503003 .
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
sac.c
|
| 3258db26ed7cedd20e1e21aca70d8304c7cef218 |
15-Jul-2015 |
kwiberg <kwiberg@webrtc.org> |
Split iSAC encoder/decoder: Test more cases (and make sure they work)
This patch tests separate iSAC encoder and decoder in more cases (32
kHz in addition to 16 kHz, and 30 ms adaptive and 60 ms nonadaptive).
In order to handle 32 kHz adaptive, the decoder needs to be told of
the encoder's sample rate (16 kHz worked already because that's the
default). And since we can't set the encoder's frame size without also
setting its bit rate, we need a way to set the decoder's bit rate as
well.
It turned out to be way too messy to continue verifying that the
bandwidth estimator does something reasonable in all these cases,
because it seems it doesn't. So the GetSetBandwidthInfo is now just
responsible for ensuring that split encoder/decoder behaves the same
as conjoined encoder/decoder; the job of verifying that the bandwidth
estimator does its job properly falls on some other test (that doesn't
exist yet).
Review URL: https://codereview.webrtc.org/1225093005
Cr-Commit-Position: refs/heads/master@{#9583}
sac.c
|
| 2224294c52fc0bb95eedb8187a62bedf1b5eb853 |
03-Jul-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
iSAC: Functions for importing and exporting bandwidth est. info
They make it possible to send bandwidth estimation info from decoder
to encoder even if they are separate objects (which we want them to be
because multithreading).
R=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1208923002.
Cr-Commit-Position: refs/heads/master@{#9535}
andwidth_estimator.c
andwidth_estimator.h
sac.c
tructs.h
|
| f4eca64596f8d29eeaedd9365177d8004a0cac4a |
02-Jul-2015 |
kwiberg <kwiberg@webrtc.org> |
iSAC: Pad with zeros instead of random data, to make testing easier
Using random "garbage" bytes makes testing harder for no good reason.
Any deterministic sequence would do, but we choose all zeros because
it's simple.
Review URL: https://codereview.webrtc.org/1211243014
Cr-Commit-Position: refs/heads/master@{#9532}
sac.c
|
| 1d34fe979c52e5826c5c8162759b0167b2607836 |
16-Jun-2015 |
henrika <henrika@chromium.org> |
Adds support for webrtc::test::ResourcePath on iOS
BUG=webrtc:4752
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1178843002.
Cr-Commit-Position: refs/heads/master@{#9445}
sac_unittest.cc
|
| ac81163011c586fda1e74fd9d53a7156856dfd8c |
16-Jun-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
iSAC: Move global trig tables into the codec instance
These tables are constant, so it makes sense for all encoders to share
one copy---but it was initialized in a racy way, and there's no
appealing way to fix that without adding dependencies on locking
functions. So we simply give each codec instance its own copy, which
costs 8 * (240 + 240 + 120 + 120) = 5760 bytes apiece.
As noted in the TODO comment, the size of the tables could be reduced,
and they could be filled in at compile-time, but that would make the
encoder output slightly different, which would mess with our tests.
R=henrik.lundin@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1177993003.
Cr-Commit-Position: refs/heads/master@{#9442}
odec.h
ecode.c
ncode.c
ntialize.c
sac.c
tructs.h
ransform.c
|
| a6aa6d96f8aa10736c76deb2a6ef09027d375a4a |
15-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
Fix a data race in AudioEncoderMutableImpl and derived classes
Before this change, it could happen that a caller would get a pointer
to the encoder_ but not use it before another thread called the
Reconstruct method, changing the pointer. This of course resulted in
bad access crashes. With this change, each use of the pointer acquired
from the encoder() method is protected by the same lock that is
required to update the pointer. Note that this fix is probably too
aggressive, since it also affects the Opus implementation; the crash
has so far only been seen for iSAC.
Also adding a test to trigger the problem. The test did not trigger
the problem deterministically, but out would typically find it in less
than 1000 runs.
BUG=chromium:499468
R=jmarusic@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1176303004.
Cr-Commit-Position: refs/heads/master@{#9436}
udio_encoder_isac.cc
|
| aba07ef6d92bf1ded7ad1af49b54a8e6652dfcbb |
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reland "Upconvert various types to int.", isac portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24. Specifically, the
files in webrtc/modules/audio_coding/codecs/isac/ are relanded.
The original commit message is below:
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
TBR=kwiberg
Review URL: https://codereview.webrtc.org/1179093002
Cr-Commit-Position: refs/heads/master@{#9422}
rc.c
rc.h
sac.c
sac_unittest.cc
|
| 728d9037c016c01295177fa700fc7927f0bb80bb |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects.
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
andwidth_estimator.c
andwidth_estimator.h
pc_analysis.c
|
| b7e5054414ff524f9db81dab7917729b8c4c8bcb |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
sac.c
|
| cb180976dd0e9672cde4523d87b5f4857478b5e9 |
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Revert "Upconvert various types to int."
This reverts commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.
BUG=499241
TBR=hlundin
Review URL: https://codereview.webrtc.org/1179953003
Cr-Commit-Position: refs/heads/master@{#9418}
rc.c
rc.h
sac.c
sac_unittest.cc
|
| a2c79405b407162119954d57855c8c04c043df76 |
10-Jun-2015 |
henrika <henrika@chromium.org> |
Ensures that modules_unittests runs on iOS
BUG=4752
R=tkchin@chromium.org
Review URL: https://codereview.webrtc.org/1171033002.
Cr-Commit-Position: refs/heads/master@{#9408}
sac_unittest.cc
|
| 83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Upconvert various types to int.
Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t.
Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C."
This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change.
BUG=none
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54629004
Cr-Commit-Position: refs/heads/master@{#9405}
rc.c
rc.h
sac.c
sac_unittest.cc
|
| dcccab3ebb623df74fbb1425da2cb9d9a42439fa |
07-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
New interface: AudioEncoderMutable
With implementations for all codecs. It has no users yet. This new
interface is the same as AudioEncoder (in fact it is a subclass) but
it allows changing some parameters after construction.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51679004
Cr-Commit-Position: refs/heads/master@{#9149}
udio_encoder_isac.cc
|
| d3e8eda8398640fac1b51f6f986f4fbb3fb5dd21 |
23-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
(Re-land) AudioEncoderDecoderIsac: Merge the two config structs
This reverts commit 599beb86, which in turn reverted 7c324cac. What
makes it work this time is that we don't remove the option of setting
bit_rate to 0 in order to ask for the default value.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4228, chromium:478161
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48199004
Cr-Commit-Position: refs/heads/master@{#9068}
udio_encoder_isac_unittest.cc
|
| 6069032ebb9026bc22ae18d1c4b8e8315140a907 |
20-Mar-2015 |
bjornv@webrtc.org <bjornv@webrtc.org> |
Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
ntropy_coding.c
|
| 0c5b137e7eca6afcc1737ed66fad4302f43dad92 |
13-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Remove support for iSAC RCU
The current way that iSAC RCU is packetized and sent as a RED packet,
with the same payload type for primary and redundant payloads, does
not follow the specification for RED. As it is now, it is impossible
for a receiver to know if an incoming RED packet with iSAC payloads
inside consists of two "primary" (but time-shifted) payloads, or one
primary and one RCU payload. The RED standard stipulates that the
former option is the correct interpretation, while our implementation
currently applies the latter.
This CL removes support for iSAC RCU from Audio Coding Module, but
leaves it in the iSAC codec itself (i.e., in the C implementation).
BUG=4402
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45569004
Cr-Commit-Position: refs/heads/master@{#8713}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8713 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac_red_unittest.cc
|
| 51ccf376387266225cd8c78e63238b725860f0af |
10-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: add method MaxEncodedBytes
Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call.
Unit tests were updated to use the new method.
Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation.
Other refactoring work that was done, that may not be obvious why:
1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive().
2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40259005
Cr-Commit-Position: refs/heads/master@{#8671}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac_red_unittest.cc
|
| 1d25c8719989fdbd30a845921e5bf0bd60535ad8 |
04-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
This effectively reverts r8578.
TBR=jmarusic@webrtc.org
Original commit message:
Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
With this change, support for iSAC-RED is incorporated into the
regular AudioEncoderDecoderIsac class.
COAUTHOR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44539004
Cr-Commit-Position: refs/heads/master@{#8589}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8589 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
udio_encoder_isac_red_unittest.cc
|
| bcef431902ad4addbd8fce8b1289b35cdfec1416 |
03-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert r8577 "Collapse AudioEncoderDecoderIsacRed into ..."
Some of the build bots seems to have reacted to this change.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42169004
Cr-Commit-Position: refs/heads/master@{#8578}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8578 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
udio_encoder_isac_red_unittest.cc
|
| 1fc28f2305947dda3a8aa83cb86190af3edd16fb |
03-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Collapse AudioEncoderDecoderIsacRed into AudioEncoderDecoderIsac
With this change, support for iSAC-RED is incorporated into the regular
AudioEncoderDecoderIsac class.
COAUTHOR=kwiberg@webrtc.org
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43549004
Cr-Commit-Position: refs/heads/master@{#8577}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8577 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
udio_encoder_isac_red_unittest.cc
|
| b1f0de30be3397eba3d423b71abc5c50db2a1665 |
26-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: change Encode and EncodeInternal return type to void
After code cleanup done on issues:
https://webrtc-codereview.appspot.com/34259004/
https://webrtc-codereview.appspot.com/43409004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/34309004/
https://webrtc-codereview.appspot.com/36209004/
https://webrtc-codereview.appspot.com/40899004/
https://webrtc-codereview.appspot.com/39279004/
https://webrtc-codereview.appspot.com/42099005/
and the similar work done for AudioEncoderDecoderIsacT, methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38279004
Cr-Commit-Position: refs/heads/master@{#8518}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac_red_unittest.cc
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac_red_unittest.cc
|
| d4dfba8ea1efacb88de636b466140b0176a32097 |
25-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
iSAC Decode: Prevent Memcheck from complaining about uninitialized value
Without this patch, Valgrind's Memcheck was complaining that the test
for whether we should return -1 following the call to
WebRtcIsac_DecodeLb made a conditional branch or move based on the
value of numSamplesLB, which was uninitialized if WebRtcIsac_DecodeLb
failed.
However, as can be seen in the source, the control flow only depends
on the value of numSamplesLB if numDecodedBytesLB >= 0; i.e., if
WebRtcIsac_DecodeLb returned successfully, in which case numSamplesLB
is always initialized. The discrepancy is due to the fact that
Valgrind works on the generated machine code, which contains spurious
such dependencies. The generated code for this test:
if ((numDecodedBytesLB < 0) || (numDecodedBytesLB > lenEncodedLBBytes) ||
(numSamplesLB > MAX_FRAMESAMPLES)) {
looks like this:
95: 0f bf 45 d6 movswl -0x2a(%rbp),%eax
99: 3d c0 03 00 00 cmp $0x3c0,%eax
9e: 0f 8f 45 01 00 00 jg 1e9 <Decode+0x1e9>
a4: 44 89 f0 mov %r14d,%eax
a7: c1 e0 10 shl $0x10,%eax
aa: 0f 88 39 01 00 00 js 1e9 <Decode+0x1e9>
b0: 41 0f bf ce movswl %r14w,%ecx
b4: 89 8d 98 e1 ff ff mov %ecx,-0x1e68(%rbp)
ba: 41 0f bf c7 movswl %r15w,%eax
be: 39 c1 cmp %eax,%ecx
c0: 0f 8f 23 01 00 00 jg 1e9 <Decode+0x1e9>
Note how the compiler has seemingly ignored the C language's guarantee
that the arguments to || must be evaluated in left-to-right order, and
compares numSamplesLB (%eax) with MAX_FRAMESAMPLES (0x3c0, a.k.a. 960)
before the other two conditions! If the uninitialized value in
numSamplesLB happens to be greater than 960, we'll jump to
Decode+0x1e9 (where we'll return -1) without even looking at the other
two conditions. Has the compiler generated broken code?
Well, no. If numDecodedBytesLB is < 0 so that numSamplesLB is
uninitialized, we'll end up jumping to 1e9 whether that value is
greater than 960 or not; we'll just do it with different jump
instructions. This is entirely invisible as far as the C language is
concerned, but the dependency on the uninitialized value is visible at
the machine code level, which is why Memcheck complains.
This patch solves the problem by pragmatically initializing
numSamplesLB before the call even though it isn't necessary other than
for placating Memcheck.
BUG=4143
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36309004
Cr-Commit-Position: refs/heads/master@{#8492}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8492 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| e35fa96cbeabeea49aaf6ecb3c3fce154ad5ab2a |
16-Feb-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Move isacfix.gypi and isac.gypi
Move webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
and webrtc/modules/audio_coding/codecs/isac/main/source/isac.gypi to
webrtc/modules/audio_coding/codecs/isac to pass recently
added _CheckNoSourcesAboveGyp presubmit rule.
BUG=4002
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37269004
Cr-Commit-Position: refs/heads/master@{#8376}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8376 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| bdebccf384db689931ba3df9afbcd59c85ddb8e2 |
29-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Fix a number of things in AudioEncoderDecoderIsac*
- Add max_bit_rate and max_payload_size_bytes to config structs.
- Fix support for 48 kHz sample rate.
- Fix iSAC-RED.
- Add method UpdateDecoderSampleRate().
- Update locking structure with a separate lock for local member
variables used by the encoder methods.
BUG=3926
COAUTHOR:kwiberg@webrtc.org
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41659004
Cr-Commit-Position: refs/heads/master@{#8204}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8204 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac_red_unittest.cc
|
| eed7a22bbfd8906d3ffd47c9a7c7f7f9308172d7 |
18-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Make an AudioEncoder subclass for iSAC redundant encoding
Adding unit test, too.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
udio_encoder_isac_red_unittest.cc
|
| eb544460e47140d494dddf1217a698a1dcf4dee0 |
17-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rename _t struct types in audio_coding.
_t names are reserved in POSIX.
R=henrik.lundin@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/34509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
odec.h
ncode.c
ntropy_coding.c
ntropy_coding.h
tructs.h
|
| c1c9291e9b73c3a9633cd5c5cbc2e243ab5d6920 |
16-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Make an AudioEncoder subclass for RED
This class only supports the simple case of payload duplication. That
is, one single encoder is used, and the redundant payload is a one-step
delayed payload.
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7913 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| 88bdec8c3ab768c9cdb31b4216f82dba815ea516 |
16-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
AudioEncoder subclass for iSACfix
This patch refactors AudioEncoderDecoderIsac so that it can share
almost all code with the very similar AudioEncoderDecoderIsacFix.
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29259004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7912 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
sac.gypi
|
| 3b79daff14127f3adb19b16d94336d44ff49e841 |
12-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Moving encoded_bytes into EncodedInfo
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
|
| b3ad8cf6ca8d2d9fa1152bb8bff624c0be83373f |
11-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Make an AudioEncoder subclass for iSAC
BUG=3926
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
sac.gypi
|
| 1431e4dd1c0870971afde9456702d4b76aca3324 |
11-Nov-2014 |
turaj@webrtc.org <turaj@webrtc.org> |
Revert 7675 "Make an AudioEncoder subclass for iSAC"
Above CL did not compile on Android. Followings are links to Android builds
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369
http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320
> Make an AudioEncoder subclass for iSAC
>
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/25019004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
sac.gypi
|
| 05feff013e6faf1ba7328bbf3932c90cb3de6705 |
11-Nov-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Make an AudioEncoder subclass for iSAC
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
udio_encoder_isac.cc
sac.gypi
|
| 2abebe7bafe3887ed20756031beb2b514313d0c6 |
20-Oct-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >>
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
ntropy_coding.c
|
| 396a5e00012ea505a58447e9378a64227e74346b |
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.
BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| 3f7f899a15c2685a8e45484f7b2c540771d28d90 |
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_UpdateBwEstimate et al.: Type byte streams as uint8, not uint16
This patch changes the signature of WebRtcIsac_UpdateBwEstimate,
WebRtcIsacfix_UpdateBwEstimate, and WebRtcIsacfix_UpdateBwEstimate1 so
that they expect the encoded data to be uint8 arrays, not uint16,
which is more natural. The implementations of the functions are left
unchanged for now.
BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7430 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
sac_unittest.cc
|
| 1172988c794d15706b4c951dcbaa57b11221d225 |
13-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
The affected functions are
WebRtcIsacfix_ReadFrameLen
WebRtcIsacfix_GetNewBitStream
WebRtcIsacfix_ReadBwIndex
and
WebRtcIsac_ReadFrameLen
WebRtcIsac_GetNewBitStream
WebRtcIsac_ReadBwIndex
WebRtcIsac_GetRedPayload
BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22979004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| 7c15510f389b00fea03e8512cf1a09d0a344b8e9 |
29-Sep-2014 |
bjornv@webrtc.org <bjornv@webrtc.org> |
common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary.
Affects both fixed and float point versions of iSAC
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 4adac7df-926f-26a2-2b94-8c16560cd09d
rc.c
|
| 7ee24a79065a655dcc62a27fd22e0cc77fee6d68 |
24-Sep-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7266
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
sac_unittest.cc
|
| a3c4d4dd2cece2cfbbd687eb76da833c37fbde3c |
23-Sep-2014 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795
> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
>
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
>
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19229004
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
sac_unittest.cc
|
| 8c5740b48507e8fbb2c56c7dd52a1197ebb5d20d |
23-Sep-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
We have to fix both at once, since there's a macro that calls one of
them or the other.
BUG=909
R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7266 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
sac_unittest.cc
|
| 262e676a08fc29ee6c414f5858d68697be983515 |
04-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
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
rc.h
ncode_lpc_swb.c
ncode_lpc_swb.h
pc_gain_swb_tables.c
pc_gain_swb_tables.h
pc_shape_swb12_tables.c
pc_shape_swb12_tables.h
pc_shape_swb16_tables.c
pc_shape_swb16_tables.h
s_specific_inline.h
itch_gain_tables.h
itch_lag_tables.h
|
| 1b8b4c4959c5a1cf08af527e28eef86940d73880 |
03-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
rc.h
ncode_lpc_swb.c
ncode_lpc_swb.h
pc_gain_swb_tables.c
pc_gain_swb_tables.h
pc_shape_swb12_tables.c
pc_shape_swb12_tables.h
pc_shape_swb16_tables.c
pc_shape_swb16_tables.h
s_specific_inline.h
itch_gain_tables.h
itch_lag_tables.h
|
| 9730d3aae91799334dea86a0439f86fa7c4ab2a5 |
03-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
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
rc.h
ncode_lpc_swb.c
ncode_lpc_swb.h
pc_gain_swb_tables.c
pc_gain_swb_tables.h
pc_shape_swb12_tables.c
pc_shape_swb12_tables.h
pc_shape_swb16_tables.c
pc_shape_swb16_tables.h
s_specific_inline.h
itch_gain_tables.h
itch_lag_tables.h
|
| 524b8f7304090eec09e4251ca234e0ea083ce858 |
31-Aug-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
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
ilter_functions.c
attice.c
itch_estimator.c
|
| 047a46f8b49e7100d7727377c89f109542125b9c |
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove Android.mk build files.
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.
R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
ndroid.mk
|
| eec6ecdb1e249871dd25d04b62fc9ddc03dc8a34 |
11-Jul-2014 |
tommi@webrtc.org <tommi@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org, pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| 2c89b5cb27536eac2ca298c4a36f3a5ccb903141 |
14-Apr-2014 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
WNERS
|
| e2fc13e42f3881d2833af627a23ac77e938515ec |
15-Feb-2014 |
bjornv@webrtc.org <bjornv@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactoring common_audio/signal_processing: Removed two macros used by isac only.
Removed a macro for malloc() and one for free(). They are only used by the audio codec isac, where I replaced the macro with its implementation.
Further, the includes were updated with full paths and put in alphabetical order.
BUG=N/A
TESTED=trybots,module_tests,module_unittests
R=turaj@webrtc.org, turajs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5554 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| a80be4b23c617a070d1ba7a133e948d337aabe4b |
11-Feb-2014 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add boundary checking to supress gcc 4.8.3 warning.
BUG=2888
Test=try, voe_cmd_test
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5526 4adac7df-926f-26a2-2b94-8c16560cd09d
ntropy_coding.c
pectrum_ar_model_tables.c
pectrum_ar_model_tables.h
|
| 4ffd9c74233443a3bedb10f92bd735c2c7ef262c |
16-Jan-2014 |
aluebs@webrtc.org <aluebs@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add full path to headers
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5395 4adac7df-926f-26a2-2b94-8c16560cd09d
tructs.h
|
| 60730cfe3ce80e4023cd678373456cb703f000a4 |
07-Jan-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the requirement to call set_sample_rate_hz and friends.
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| 57eb8586986a2c77b99124c270bc6caa11165f7f |
11-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ".." from include_dirs in build/common.
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2332004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| 621df678c8690f36875b0b34d45393df58662172 |
22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
|
| 137b3793d9a73adc4b730a8ed18a7b009ea4aa59 |
27-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Only use -lm on Linux in ISAC.
Remove unneeded WEBRTC_LINUX define.
BUG=crbug.com/298656
TESTED=Passed trybots.
R=wjia@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2313004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4865 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
s_specific_inline.h
|
| 76fe9309b9f532d21d889b06f722d57e0139e9d0 |
19-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use link_settings instead of all_dependent_settings to pacify xcode gyp generator
Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2261004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| ccddd0a9417e5c015f0f4f6a75e1179fe33514d7 |
19-Sep-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Roll webrtc's chromium_revision 217707:224141
Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2177007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| 4ce838934cdbfa1777aa76237484f64d00e0281e |
14-May-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Address sanitizer out of bounds read in iSAC
BUG=issue1770
TBR=tlegrand@google.com
Review URL: https://webrtc-codereview.appspot.com/1472006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.c
sac_unittest.cc
|
| 342353780d411d1fd9eed5cd5e42782100a1d724 |
01-May-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Consolidate common_audio into a single target.
In principle should reduce gyp processing time, but the difference was not measurable. In any case, it's a good simplification that aligns with having a single common_video target.
R=bjornv@webrtc.org, kma@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3928 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| 6e788df19ef1e37049717757218fe1e74bbce1c2 |
16-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove vim/emacs modelines from .gypi files
BUG=1655
Review URL: https://webrtc-codereview.appspot.com/1326005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| 0946a56023d821e0deca04029bb016ae1f23aa82 |
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t etc. in audio_coding/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1271006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
rith_routines.c
rith_routines.h
rith_routines_hist.c
rith_routines_logist.c
andwidth_estimator.c
andwidth_estimator.h
odec.h
rc.c
rc.h
ecode.c
ecode_bwe.c
ncode.c
ncode_lpc_swb.c
ncode_lpc_swb.h
ntropy_coding.c
ntropy_coding.h
sac.c
pc_analysis.c
pc_analysis.h
pc_gain_swb_tables.c
pc_gain_swb_tables.h
pc_shape_swb12_tables.c
pc_shape_swb12_tables.h
pc_shape_swb16_tables.c
pc_shape_swb16_tables.h
pc_tables.c
pc_tables.h
itch_gain_tables.c
itch_gain_tables.h
itch_lag_tables.c
itch_lag_tables.h
pectrum_ar_model_tables.c
pectrum_ar_model_tables.h
tructs.h
ransform.c
|
| 09738616de34a33ba97c2c5d784f537166adc244 |
22-Jan-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes payload spelling error.
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/1052006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3398 4adac7df-926f-26a2-2b94-8c16560cd09d
tructs.h
|
| a3c82bf6673a2e0367bcb89a287cdc9ec0c37a53 |
19-Jan-2013 |
wjia@webrtc.org <wjia@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove '<(library)' in gyp files.
This will remove all usage of '<(library)' in all webrtc gyp files.
Review URL: https://webrtc-codereview.appspot.com/1049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
sac.gypi
|
| b0dff12d2bfd2be52c07b0bcce5a36938ea4f491 |
03-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
48 kHz extension to iSAC.
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
ecode_bwe.c
sac.c
ettings.h
tructs.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
ndroid.mk
rith_routines.c
rith_routines.h
rith_routines_hist.c
rith_routines_logist.c
andwidth_estimator.c
andwidth_estimator.h
odec.h
rc.c
rc.h
ecode.c
ecode_bwe.c
ncode.c
ncode_lpc_swb.c
ncode_lpc_swb.h
ntropy_coding.c
ntropy_coding.h
ft.c
ft.h
ilter_functions.c
ilterbank_tables.c
ilterbank_tables.h
ilterbanks.c
ntialize.c
sac.c
sac.gypi
attice.c
pc_analysis.c
pc_analysis.h
pc_gain_swb_tables.c
pc_gain_swb_tables.h
pc_shape_swb12_tables.c
pc_shape_swb12_tables.h
pc_shape_swb16_tables.c
pc_shape_swb16_tables.h
pc_tables.c
pc_tables.h
s_specific_inline.h
itch_estimator.c
itch_estimator.h
itch_filter.c
itch_gain_tables.c
itch_gain_tables.h
itch_lag_tables.c
itch_lag_tables.h
ettings.h
pectrum_ar_model_tables.c
pectrum_ar_model_tables.h
tructs.h
ransform.c
|