Commit Graph

306 Commits

Author SHA1 Message Date
dce40cf804 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}
2015-08-24 21:52:45 +00:00
608c3cfe77 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}
2015-08-24 09:03:28 +00:00
805d8fb6eb 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}
2015-08-14 19:38:09 +00:00
b3cc77f4ba Re-enable WebRtcIsacfix_AllpassFilter2FixDec16Neon
WebRtcIsacfix_AllpassFilter2FixDec16Neon was disabled due to a Clang
bug. The bug is fixed in current Clang version, re-enable it in this patch.

BUG=4567
R=andrew@webrtc.org, kjellander@webrtc.org
TEST=buildbot build

Change-Id: I71e309cec6caf376181cf9c299c9e8967c9a328e

Review URL: https://codereview.webrtc.org/1194773002 .

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#9645}
2015-07-28 03:18:19 +00:00
b297c5a01f 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}
2015-07-22 22:17:26 +00:00
3258db26ed 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}
2015-07-15 01:54:43 +00:00
2224294c52 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}
2015-07-03 02:04:46 +00:00
f4eca64596 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}
2015-07-02 09:10:11 +00:00
c6891248b5 Simplify OWNERS structure in modules/audio_coding
All ownership is now handled by the top-level OWNERS file in
modules/audio_coding.

NOTRY=True

Review URL: https://codereview.webrtc.org/1212133005

Cr-Commit-Position: refs/heads/master@{#9512}
2015-06-29 11:54:50 +00:00
ac4234ccfc Add a [rtc_]build_with_neon variable to unify conditions.
Also consolidate ARM options for gn in an arm_neon_config.

R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org

Review URL: https://codereview.webrtc.org/1181373004.

Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-25 01:25:59 +00:00
3e89dbf458 Add AudioEncoder::GetTargetBitrate
The GetTargetBitrate implementation will return the
target bitrate of the codec. This may differ from the
desired target bitrate, as set by SetTargetBitrate, depending on implementation.

Tests are updated to exercise the new functionality.

R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1184313002.

Cr-Commit-Position: refs/heads/master@{#9461}
2015-06-18 12:58:46 +00:00
1d34fe979c 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}
2015-06-16 08:04:24 +00:00
ac81163011 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}
2015-06-15 22:02:45 +00:00
d10cd97ad3 Make global constants 'const'
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1188533002.

Cr-Commit-Position: refs/heads/master@{#9438}
2015-06-15 13:07:11 +00:00
a6aa6d96f8 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}
2015-06-15 11:46:24 +00:00
01bbe3eb8c Fix AppRTCDemo crash under iOS armv7 devices
Fix AppRTCDemo crash under iOS due to the unaligned access in vld1
instruction in iSACFix codec, which is not allowed in iOS build.

BUG=4717
R=andrew@webrtc.org, jridges@masque.com
TEST=Run the AppRTCDemo

Change-Id: Ie5fbc9b8ae88cd00b243a8e65cab95b00362a9da

Review URL: https://codereview.webrtc.org/1182493006.

Cr-Commit-Position: refs/heads/master@{#9432}
2015-06-15 06:57:00 +00:00
bba7807078 Reland "Upconvert various types to int.", misc. codecs portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.  Specifically, the
files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as
well as webrtc/modules/audio_coding/main/test/opus_test.cc, 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/1179093003

Cr-Commit-Position: refs/heads/master@{#9424}
2015-06-12 02:02:58 +00:00
a8b335c709 Reland "Upconvert various types to int.", ilbc portion.
This reverts portions of commit cb180976dd0e9672cde4523d87b5f4857478b5e9, which
reverted commit 83ad33a8aed1fb00e422b6abd33c3e8942821c24.  Specifically, the
files in webrtc/modules/audio_coding/codecs/ilbc/ 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/1184643002

Cr-Commit-Position: refs/heads/master@{#9423}
2015-06-12 01:51:33 +00:00
aba07ef6d9 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}
2015-06-12 01:19:37 +00:00
728d9037c0 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}
2015-06-11 21:31:48 +00:00
b7e5054414 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}
2015-06-11 19:56:03 +00:00
cb180976dd 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}
2015-06-11 19:42:42 +00:00
f045e4da43 Prepare to convert various types to size_t.
This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Review URL: https://codereview.webrtc.org/1174813003

Cr-Commit-Position: refs/heads/master@{#9413}
2015-06-11 04:15:51 +00:00
786dbdcc38 Rename targets to use lower case format.
It makes writing a build script for merging libraries
across architectures easier. See talk/build/build_ios_libs.sh.

BUG=
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1171793002.

Cr-Commit-Position: refs/heads/master@{#9412}
2015-06-10 20:45:12 +00:00
a2c79405b4 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}
2015-06-10 11:24:58 +00:00
2a10087d5e Manual cleanups following clang-formatting.
This primarily addresses two things:
* Tab characters still present, mostly in comments
* printfs split across multiple lines in a suboptimal way

Along the way this fixes a few spelling errors and other minor changes.

BUG=none
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52689004

Cr-Commit-Position: refs/heads/master@{#9406}
2015-06-10 00:26:48 +00:00
83ad33a8ae 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}
2015-06-10 00:20:09 +00:00
248b0b0790 Run clang-format --style=Chromium on four files I'm otherwise touching.
The existing style in these files is pretty inconsistent and wildly divergent
from most of WebRTC/Chromium; clang-formatting them not only makes them easier
to read, it makes me see fewer presubmit errors when I try to touch the files to
make other changes.

BUG=none
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52019004

Cr-Commit-Position: refs/heads/master@{#9364}
2015-06-03 19:32:55 +00:00
323b132f5e Protect ACM decoder buffer in stereo.
In https://code.google.com/p/webrtc/source/detail?r=8730, I did a protection on ACM decoder buffer from being overflow.

However, the I misunderstood the return unit for PacketDuration(), and therefore, stereo decoders are not well protected.

This CL fixed this.

BUG=4361
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47289004

Cr-Commit-Position: refs/heads/master@{#9275}
2015-05-25 11:49:45 +00:00
367c868c99 AudioEncoderCng: Handle case where speech encoder is reset
Previously, AudioEncoderCng required the speech encoder to not change
its mind regarding the number of 10 ms frames in the next packet
between calls to AudioEncoderCng::EncodeInternal()---specifically, it
could handle an upward but not a downward adjustment. With this patch,
it can handle a downward adjustment too, by simply saving the
overshoot data for the next call to EncodeInternal().

It will still not handle the case where the encoder's reported number
of 10 ms frames in the next packet is inconsistent with the behavior
of its Encode() function when called with no intervening changes to
the encoder.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53469005

Cr-Commit-Position: refs/heads/master@{#9261}
2015-05-22 13:13:24 +00:00
cb7f8ce2df Clear ARM NEON flag
Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one
WEBRTC_HAS_NEON.
Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON.
Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980

Review URL: https://webrtc-codereview.appspot.com/49309004

Cr-Commit-Position: refs/heads/master@{#9228}
2015-05-20 05:20:04 +00:00
7e0c7d49ea Add support for external encoders in ACM
Also introduce tests using external (mock) encoders, both for
CodecOwner and for AudioCodingModule.

Support for external decoders is still missing.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4474
R=jmarusic@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49939004

Cr-Commit-Position: refs/heads/master@{#9206}
2015-05-18 12:52:13 +00:00
092041c1cd Setting OPUS_SIGNAL_VOICE when enable DTX.
A better solution than forcing OPUS_APPLICATION_VOIP when enabling DTX has been found, which is to set OPUS_SIGNAL_VOICE.

This reduces the uncertainty of entering DTX over silence period of audio.

This CL contains the setup of OPUS_SIGNAL_VOICE and decoupling opus application mode with DTX.

BUG=4559
R=henrik.lundin@webrtc.org, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46959004

Cr-Commit-Position: refs/heads/master@{#9168}
2015-05-11 10:19:36 +00:00
dcccab3ebb 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}
2015-05-07 10:35:18 +00:00
f242e665b4 Replace asm NEON function by intrinsics implementation on ARMv7
Passed building isac_neon and modules_unittests on Android ARMv7.
Passed modules_unittests with following filters:
  --gtest_filter=FiltersTest*
  --gtest_filter=LpcMaskingModelTest*
  --gtest_filter=TransformTest*
  --gtest_filter=FilterBanksTest*

WebRtcIsacfix_CalculateResidualEnergyNeon is removed, refer more in
Issue 4224.

The old review url is at: https://webrtc-codereview.appspot.com/37259004/

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48319005

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Change-Id: I4c16e15930f1b3449d67b67bf023fac28121dff8
Cr-Commit-Position: refs/heads/master@{#9140}
2015-05-06 08:39:37 +00:00
589699eea2 Fix bug in transform_neon.c in iSAC codec.
The bug causes AcmReceiverBitExactness and AcmSenderBitExactness test
failed in modules_unittests.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Change-Id: I18b00056c53cf4158c186d449e5ab785065cca94

Review URL: https://webrtc-codereview.appspot.com/49889004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#9138}
2015-05-06 02:25:20 +00:00
88de4792d0 AudioEncoderIsac: Print error code if CHECK for successful encoding fails
This will hopefully make the crash in bug 4577 easier to understand if
it happens again.

BUG=4577
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52389004

Cr-Commit-Position: refs/heads/master@{#9100}
2015-04-28 13:43:43 +00:00
e8a197bd07 Enable isac NEON building on Aarch64
Passed building isac_neon and modules_unittests on Android ARM64 and
ARMv7.
Passed modules_unittests with following filters:
--gtest_filter=FiltersTest*
--gtest_filter=LpcMaskingModelTest*
--gtest_filter=TransformTest*
--gtest_filter=FilterBanksTest*

WebRtcIsacfix_CalculateResidualEnergyNeon is not enabled due to Issue
4224.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44229004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

Cr-Commit-Position: refs/heads/master@{#9092}
2015-04-28 06:42:04 +00:00
d3e8eda839 (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}
2015-04-23 12:06:46 +00:00
92f9eacd13 g722 and red encoders: Use rtc::Buffer instead of scoped_ptr<uint8_t[]>
It's a win for red, and a toss-up for g722 since it never resizes its
buffer.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45219005

Cr-Commit-Position: refs/heads/master@{#9067}
2015-04-23 11:53:02 +00:00
8f85dbcce4 Reduce the number of registers used in MIPS optimizations.
This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer
flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes
compile error for MIPS, as some MIPS optimization blocks use maximum possible
number of available registers.
Also, this change contains minor GN build fix for MIPS platform regarding the
pitch_filter_mips.c / pitch_filter_c.c file inclusion.

BUG=477749
R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48139004

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

Cr-Commit-Position: refs/heads/master@{#9047}
2015-04-21 23:52:26 +00:00
599beb8687 Revert "AudioEncoderDecoderIsac: Merge the two config structs"
Reason for revert - breaks Hangouts

This reverts commit 7c324cac50ac38122b3f3b26455bc55ad834bfc0.

BUG=chromium:478161

Review URL: https://webrtc-codereview.appspot.com/43209004

Cr-Commit-Position: refs/heads/master@{#9030}
2015-04-17 21:13:59 +00:00
7c324cac50 AudioEncoderDecoderIsac: Merge the two config structs
This patch merges the Config and ConfigAdaptive structs, so that iSAC
has just one config struct like the other codecs. Future CLs will make
use of this.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45979004

Cr-Commit-Position: refs/heads/master@{#9015}
2015-04-16 04:00:18 +00:00
f2822edf61 Refactor audio_coding/codecs/isac/fix: Removed usage of macro WEBRTC_SPL_MUL_16_16_RSFT
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))

where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
- removed commented code lines used during development
- excluded fft.c since there are neon optimizations used and a removal may cause a performance regression

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

Review URL: https://webrtc-codereview.appspot.com/48799004

Cr-Commit-Position: refs/heads/master@{#8967}
2015-04-10 06:06:46 +00:00
d417c93c10 Remove android_webview_build conditions.
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.

BUG=chromium:440793
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44119005

Patch from Richard Coles <torne@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8963}
2015-04-09 15:36:13 +00:00
d4e75016a3 Refactor audio_coding/codecs/isac/fix: Removed usage of trivial macro WEBRTC_SPL_LSHIFT_W32()
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
hence trivial.
The macro name may in fact mislead the user to assume a cast/truncation to int32_t is done.
- Removing usage of it.
- Some style changes.

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

Review URL: https://webrtc-codereview.appspot.com/46749005

Cr-Commit-Position: refs/heads/master@{#8918}
2015-04-02 04:59:44 +00:00
f809b9b38d Fix bug in WebRtcIsacfix_FilterMaLoopNeon.
Pass content_browsertests in Chromium. Performance test result (lower is
better):
C version: 100%
old intrinsics Neon version (with bug): 16.5%
new intrinsics Neon version: 18.0%
asm Neon version: 23.3%

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ia0a96ac237216b635fc528f67d39319cdf246281

Review URL: https://webrtc-codereview.appspot.com/46739004

Cr-Commit-Position: refs/heads/master@{#8907}
2015-04-01 09:43:22 +00:00
582f80e95c Clamp decoder sample rate to 32000 in iSAC
We want to crate the illusion that iSAC supports 48000 Hz decoding,
while in fact it outputs 32000 Hz. This is the iSAC fullband mode.

Currently this is (also) handled by higher layers, but in future
changes this will not be the case.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47809004

Cr-Commit-Position: refs/heads/master@{#8889}
2015-03-30 13:01:47 +00:00
09b6ff9460 Disable PLC for iSAC
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).

BUG=4423
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42849004

Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-23 12:24:14 +00:00
6069032ebb 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
2015-03-20 07:03:41 +00:00