86 Commits

Author SHA1 Message Date
kwiberg
529662a44c Move array_view.h to webrtc/api/
We use ArrayView in our public API, so its header should be in
webrtc/api/.

BUG=none

Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
2017-09-04 12:43:17 +00:00
minyue-webrtc
bf94fda1e4 Renaming probing_interval to bwe_period globally.
This is a follow up of https://codereview.webrtc.org/2888893002/.

Bug: None
Change-Id: Ia76903858c0a6f2801f14878980e18ae6d3b85e6
Reviewed-on: https://chromium-review.googlesource.com/646020
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li (BackIn2018March) <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19644}
2017-09-02 05:00:01 +00:00
kwiberg
ee89e7870c Replace CHECK(x && y) with two separate CHECK() calls
That way, the debug printout will tell us which of x and y that was false.

BUG=none

Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
2017-08-10 00:22:01 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
2017-07-06 19:11:40 +00:00
Henrik Kjellander
dca1e09db7 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"
This reverts commit c8fa692ec44fd6ba4fa3d085ac3161a262fc18c5.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2964773002 .
Cr-Commit-Position: refs/heads/master@{#18872}
2017-07-01 14:42:25 +00:00
kjellander
c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00
ossu
eb1fde4a26 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
2017-05-02 13:46:30 +00:00
elad.alon
dadb4dc3c9 Allow ANA to receive RPLR (recoverable packet loss rate) indications
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
2017-03-23 22:29:50 +00:00
michaelt
566d820e00 Update smoothed bitrate.
BUG=webrtc:6443

Review-Url: https://codereview.webrtc.org/2546493002
Cr-Commit-Position: refs/heads/master@{#16036}
2017-01-12 18:17:38 +00:00
minyue
4b9a2cb0d8 Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate."
The earlier attempt of this was in
https://codereview.webrtc.org/2411613002/

It was reverted due to failures on internal bots, showing that we cannot deprecate one method.

BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2538493006
Cr-Commit-Position: refs/heads/master@{#15333}
2016-11-30 14:50:08 +00:00
minyue
e69b46863a Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ )
Reason for revert:
internal bot failure

Original issue's description:
> Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
>
> BUG=webrtc:6303
>
> Committed: https://crrev.com/84e56d576806635c966093d5421c5d04c9b90746
> Cr-Commit-Position: refs/heads/master@{#15310}

TBR=kwiberg@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2537243004
Cr-Commit-Position: refs/heads/master@{#15312}
2016-11-30 09:19:06 +00:00
minyue
84e56d5768 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2411613002
Cr-Commit-Position: refs/heads/master@{#15310}
2016-11-30 08:28:07 +00:00
kwiberg
352444fcac RTC_[D]CHECK_op: Remove superfluous casts
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.

It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
2016-11-28 23:59:03 +00:00
kwiberg
af476c737f RTC_[D]CHECK_op: Remove "u" suffix on integer constants
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.

NOPRESUBMIT=true
BUG=webrtc:6645

Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
2016-11-28 23:21:51 +00:00
ossu
0eb19602a3 ComfortNoise: Calculate used scale factor in Q13
BUG=chromium:666518

Review-Url: https://codereview.webrtc.org/2519873003
Cr-Commit-Position: refs/heads/master@{#15189}
2016-11-22 13:15:29 +00:00
Henrik Kjellander
b4af3d673a Remove all references to GYP
Remove all .gyp and .gypi files.
Remove entries from OWNERS files for *.isolate, *.gyp, *.gypi
Remove unused scripts in webrtc/build.

BUG=webrtc:6323
R=henrika@webrtc.org, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/2509703002 .

Cr-Commit-Position: refs/heads/master@{#15107}
2016-11-16 19:11:38 +00:00
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
kwiberg
3f81fcd2e8 Don't recreate the speech encoder if we don't have to
If the specification for the speech encoder hasn't changed, we should
reuse it instead of recreating it. Otherwise, we lose its state. (This
problem was originally discovered because AudioEncoderOpus instances
would forget that they were supposed to be using DTX.)

BUG=webrtc:6020, chromium:622647

Review-Url: https://codereview.webrtc.org/2089183002
Cr-Commit-Position: refs/heads/master@{#13273}
2016-06-23 10:58:45 +00:00
kwiberg
4485ffb58d #include "webrtc/base/constructormagic.h" where appropriate
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.

Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1917043005

Cr-Commit-Position: refs/heads/master@{#12509}
2016-04-26 15:14:48 +00:00
ossu
97ba30eedf Convert CNG into C++ and remove it from AudioDecoder
Broke out CNG from AudioDecoder as they didn't really share an interface.

Converted the CNG code to C++, to make initialization and resource handling easier. This includes several changes to the behavior, favoring RTC_CHECKs over returning error codes.

Review URL: https://codereview.webrtc.org/1868143002

Cr-Commit-Position: refs/heads/master@{#12491}
2016-04-25 14:56:05 +00:00
ossu
2903ba5ff3 Reland Remove the deprecated EncodeInternal interface from AudioEncoder
Remove the deprecated EncodeInternal interface from AudioEncoder

Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1881003003

Cr-Commit-Position: refs/heads/master@{#12409}
2016-04-18 13:14:42 +00:00
kwiberg
6ca0a31708 We no longer use compilers that can't =default move construction and assignment
Review URL: https://codereview.webrtc.org/1891483006

Cr-Commit-Position: refs/heads/master@{#12376}
2016-04-15 12:25:03 +00:00
ossu
164bc4bbd3 Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ )
Reason for revert:
Broke import. Implementations of the old interface still exists somewhere.

Original issue's description:
> Remove the deprecated EncodeInternal interface from AudioEncoder
>
> Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
>
> BUG=webrtc:5591
>
> Committed: https://crrev.com/5222d315dbea8f3563c100cc9f2451907f70b05f
> Cr-Commit-Position: refs/heads/master@{#12329}

TBR=kwiberg@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1883543002

Cr-Commit-Position: refs/heads/master@{#12330}
2016-04-12 10:58:10 +00:00
ossu
5222d315db Remove the deprecated EncodeInternal interface from AudioEncoder
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.

BUG=webrtc:5591

Review URL: https://codereview.webrtc.org/1864993002

Cr-Commit-Position: refs/heads/master@{#12329}
2016-04-12 10:31:03 +00:00
kwiberg
6030a129c0 Pass ownership of external encoders to the ACM
We want this because otherwise the ACM uses its mutex to protect an
encoder that's owned by someone else. That someone else may easily
slip up and delete or otherwise touch the encoder before making sure
that the ACM has stopped using it, bypassing the lock.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1702943002

Cr-Commit-Position: refs/heads/master@{#11909}
2016-03-08 14:01:37 +00:00
ossu
4f43fcfd49 Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.
Renamed the new variant of EncodeInternal to EncodeImpl, so that
subclasses implementing one of the EncodeInternal don't need to
explicitly contain 'using AudioEncoder::EncodeInternal' to avoid their
implementation hiding the other variant of EncodeInternal. This causes
a warning (treated as an error) when building using GCC.

Review URL: https://codereview.webrtc.org/1764583003

Cr-Commit-Position: refs/heads/master@{#11868}
2016-03-04 08:54:38 +00:00
ossu
10a029e952 Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.
For backwards compatibility, I've added kept the old interface to
Encode() and EncodeInternal and created default implementations of both
variants of EncodeInternal(), each calling the other. At least one of
the variants must be implemented in a subclass or we'll run out of stack
and explode. Would be nice if we could catch that before runtime. :/

The new interface to EncodeInternal() is protected, since it should
never be called from the outside.

Was unable to mark the old EncodeInternal() as RTC_DEPRECATED, since the
default implementaion of the new variant needs to call it to work around
old implementations. The old Encode() variant is deprecated, at least.

Added a test for backwards compatibility in audio_encoder_unittest.cc.
For the added test I broke out MockEncodeHelper from
audio_encoder_copy_red_unittest.cc and renamed it MockAudioEncoderHelper.

Review URL: https://codereview.webrtc.org/1725143003

Cr-Commit-Position: refs/heads/master@{#11823}
2016-03-01 08:41:39 +00:00
kwiberg
91d9756bc6 Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/codecs/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1696853004

Cr-Commit-Position: refs/heads/master@{#11613}
2016-02-14 09:10:08 +00:00
Peter Kasting
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
kjellander@webrtc.org
3c652b6746 modules/audio_coding: Remove some codec include dirs
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.

None of these are used downstream.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1438663003 .

Cr-Commit-Position: refs/heads/master@{#10700}
2015-11-18 22:08:46 +00:00
kwiberg
288886b2ec Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1418423010

Cr-Commit-Position: refs/heads/master@{#10535}
2015-11-06 09:21:39 +00:00
Henrik Kjellander
74640895fa 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}
2015-10-29 10:31:11 +00:00
kwiberg
8967183bf7 Simple cleanups of AudioDecoder and AudioEncoder classes
* Make sure they're all final and don't allow copying or assignment.

  * Get rid of the single-channel PCM decoder classes.

  * Move some includes from .h to .cc files where possible.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1353803002

Cr-Commit-Position: refs/heads/master@{#10021}
2015-09-22 21:06:34 +00:00
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
kwiberg
c99ebc1490 Remove AudioEncoder methods SetMaxBitrate and SetMaxPayloadSize
And the corresponding ACM methods SetISACMaxRate and
SetISACMaxPayloadSize. They were only used in tests.

Review URL: https://codereview.webrtc.org/1311533010

Cr-Commit-Position: refs/heads/master@{#9903}
2015-09-09 07:54:10 +00:00
kwiberg
3f5f1c2ad3 Change return type of AudioEncoder::SetMaxPlaybackRate to void
There's no point in returning a status code, since the max playback rate
is only a suggestion that the encoder is free to disregard.

Review URL: https://codereview.webrtc.org/1332573003

Cr-Commit-Position: refs/heads/master@{#9900}
2015-09-09 06:15:41 +00:00
kwiberg
e9e7896293 Turn webrtc::Vad into a pure virtual interface
Review URL: https://codereview.webrtc.org/1317243005

Cr-Commit-Position: refs/heads/master@{#9899}
2015-09-09 06:04:57 +00:00
kwiberg
12cfc9b4da 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}
2015-09-08 12:57:59 +00:00
Karl Wiberg
4376648df0 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}
2015-08-27 13:22:21 +00:00
Peter Kasting
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
Henrik Lundin
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
henrika
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
Peter Kasting
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
Peter Kasting
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
Peter Kasting
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
Peter Kasting
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
Zeke Chin
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
henrika
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
Peter Kasting
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