97 Commits

Author SHA1 Message Date
kwiberg
6ff045f097 Give Audio{De,En}coderIsac* an "Impl" suffix, to free up the original names
I want to publish an API for iSAC in webrtc/api/, and I want to use
the class names Audio{De,En}coderIsac{Fix,Float}.

BUG=webrtc:7835, webrtc:7841

Review-Url: https://codereview.webrtc.org/2996593002
Cr-Commit-Position: refs/heads/master@{#19381}
2017-08-17 12:31:02 +00:00
kwiberg
96d74bb933 Opus implementation of the AudioDecoderFactoryTemplate API
(This got reverted because of a problem with the Opus encoder parts.
Re-landing without changes.)

BUG=webrtc:7837

Review-Url: https://codereview.webrtc.org/2950453002
Cr-Commit-Position: refs/heads/master@{#18855}
2017-06-30 12:24:56 +00:00
kwiberg
96da0115d7 Opus implementation of the AudioEncoderFactoryTemplate API
This was previously reverted, because external projects were using the
internal webrtc::AudioEncoderOpus class and broke when it was renamed.
This re-land avoids renaming it immediately, to give those projects
time to adapt. It also has to revert some of the changes I had made to the
Config struct, since that was also used by the same external projects.

BUG=webrtc:7831

Review-Url: https://codereview.webrtc.org/2948483002
Cr-Commit-Position: refs/heads/master@{#18852}
2017-06-30 11:23:22 +00:00
solenberg
db3c9b0f72 Expose ILBC codec in webrtc/api/audio_codecs/
BUG=webrtc:7834, webrtc:7840

Review-Url: https://codereview.webrtc.org/2951873002
Cr-Commit-Position: refs/heads/master@{#18803}
2017-06-28 09:05:04 +00:00
kwiberg
1b97e26364 Don't forget to support G722 stereo decoding
https://codereview.webrtc.org/2940833002 added support for G722
decoding with the AudioDecoderFactoryTemplate API, but forgot to
support stereo.

BUG=webrtc:7839

Review-Url: https://codereview.webrtc.org/2945423003
Cr-Commit-Position: refs/heads/master@{#18761}
2017-06-26 11:19:43 +00:00
charujain
1a610f15c3 Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )
Reason for revert:
Breaking google3 projects

Original issue's description:
> Opus implementation of the AudioEncoderFactoryTemplate API
>
> Now the templated AudioEncoderFactory can create Opus encoders!
>
> BUG=webrtc:7831
>
> Review-Url: https://codereview.webrtc.org/2930243003
> Cr-Commit-Position: refs/heads/master@{#18645}
> Committed: fe1aa82c63

TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7831

Review-Url: https://codereview.webrtc.org/2947563002
Cr-Commit-Position: refs/heads/master@{#18649}
2017-06-18 09:38:58 +00:00
charujain
eb2d2d31d1 Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ )
Reason for revert:
breaking downstream projects

Original issue's description:
> Opus implementation of the AudioDecoderFactoryTemplate API
>
> BUG=webrtc:7837
>
> Review-Url: https://codereview.webrtc.org/2942733003
> Cr-Commit-Position: refs/heads/master@{#18646}
> Committed: d053fe4ab3

TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7837

Review-Url: https://codereview.webrtc.org/2944763002
Cr-Commit-Position: refs/heads/master@{#18648}
2017-06-18 09:37:17 +00:00
kwiberg
d053fe4ab3 Opus implementation of the AudioDecoderFactoryTemplate API
BUG=webrtc:7837

Review-Url: https://codereview.webrtc.org/2942733003
Cr-Commit-Position: refs/heads/master@{#18646}
2017-06-18 01:40:52 +00:00
kwiberg
fe1aa82c63 Opus implementation of the AudioEncoderFactoryTemplate API
Now the templated AudioEncoderFactory can create Opus encoders!

BUG=webrtc:7831

Review-Url: https://codereview.webrtc.org/2930243003
Cr-Commit-Position: refs/heads/master@{#18645}
2017-06-18 01:23:03 +00:00
kwiberg
b8727aebc1 G722 implementation of the AudioEncoderFactoryTemplate API
Now the templated AudioEncoderFactory can create G722 encoders!

BUG=webrtc:7833

Review-Url: https://codereview.webrtc.org/2934833002
Cr-Commit-Position: refs/heads/master@{#18644}
2017-06-18 00:41:59 +00:00
kwiberg
b1ed7f09c0 G722 implementation of the AudioDecoderFactoryTemplate API
Now the templated AudioDecoderFactory can create G722 decoders!

BUG=webrtc:7839

Review-Url: https://codereview.webrtc.org/2940833002
Cr-Commit-Position: refs/heads/master@{#18643}
2017-06-18 00:30:09 +00:00
ossu
a1a040a4a4 Injectable audio encoders: BuiltinAudioEncoderFactory
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 17:03:21 +00:00
michaelt
54340d8e75 Change opus min bitrate.
BUG=webrtc:7087

Review-Url: https://codereview.webrtc.org/2668693003
Cr-Commit-Position: refs/heads/master@{#16383}
2017-01-31 17:06:53 +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
solenberg
2779bab02a Support receiving DTMF for multiple RTP clock rates.
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2337473002
Cr-Commit-Position: refs/heads/master@{#15128}
2016-11-17 12:45:25 +00:00
nisse
c4d2dc4e02 Delete DataLog abstraction, which was almost unused.
Configuration with rtc_enable_data_logging = true was broken in cl
https://codereview.webrtc.org/2054373002/ (which deleted the
FileWrapper::WriteText method), and apparently noone noticed.

BUG=None

Review-Url: https://codereview.webrtc.org/2439473002
Cr-Commit-Position: refs/heads/master@{#14719}
2016-10-21 08:53:01 +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
c0f2dcf9ed NetEq decoder database: Don't keep track of sample rate for builtin decoders
This allows us to get rid of the function that computes it, which gets
us one step closer to getting rid of the NetEqDecoder type.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2021063002
Cr-Commit-Position: refs/heads/master@{#12974}
2016-05-31 13:28:09 +00:00
kwiberg
6c2eab34f8 AudioDecoder: New method SampleRateHz, + implementations for our codecs
This will let NetEq (and the factory, and every layer in between) keep
track of just the decoder, instead of decoder and sample rate.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2024633002
Cr-Commit-Position: refs/heads/master@{#12968}
2016-05-31 09:46:28 +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
kjellander
0206000a66 iOS: Add resource files for tests and implement OutputPath
With this change the following tests have been successfully
passing in the iOS Simulator for iPhone 5 and iOS 9:
* audio_decoder_unittests
* common_video_unittests
* modules_tests
* rtc_api_objc_tests
* rtc_pc_unittests
* system_wrappers_unittests
* voice_engine_unittests

The modules_unittests and common_audio_unittests are
handled in https://codereview.webrtc.org/1698033002/

BUG=webrtc:4755
NOTRY=True

Review URL: https://codereview.webrtc.org/1694353003

Cr-Commit-Position: refs/heads/master@{#11646}
2016-02-17 06:06:17 +00:00
kwiberg
2d0c33277c Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1697823002

Cr-Commit-Position: refs/heads/master@{#11616}
2016-02-14 17:28:39 +00:00
kwiberg
5b659c0d10 Special-case android-arm64 in codec bitexactness tests
We already had a special case for android, but it only worked for arm32.

BUG=webrtc:4198, webrtc:4199

Review URL: https://codereview.webrtc.org/1512833003

Cr-Commit-Position: refs/heads/master@{#10989}
2015-12-11 15:34:05 +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
kwiberg
ee1879ca40 Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
This operation was relatively simple, since no one was doing anything
fishy with this enum. A large number of lines had to be changed
because the enum values now live in their own namespace, but this is
arguably worth it since it is now much clearer what sort of constant
they are.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1424083002

Cr-Commit-Position: refs/heads/master@{#10449}
2015-10-29 13:20:33 +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
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
2015-10-28 17:17:50 +00:00
henrik.lundin
06b869f11a Delete iSAC-fb from NetEq
This is no longer used. Related code in the iSAC codec itself will be
deleted a follow-up CL.

BUG=4210

Review URL: https://codereview.webrtc.org/1404463003

Cr-Commit-Position: refs/heads/master@{#10272}
2015-10-14 10:44:59 +00:00
kwiberg
98ab3a46d6 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.

(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368843003

Cr-Commit-Position: refs/heads/master@{#10127}
2015-10-01 04:54:29 +00:00
solenberg
3fd7be4cb1 Revert of Don't link with audio codecs that we don't use (patchset #4 id:60001 of https://codereview.webrtc.org/1349393003/ )
Reason for revert:
Breaking Chromium FYI bots.

Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}

TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1368933002

Cr-Commit-Position: refs/heads/master@{#10069}
2015-09-25 08:36:11 +00:00
kwiberg
f66a925142 Don't link with audio codecs that we don't use
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.

(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1349393003

Cr-Commit-Position: refs/heads/master@{#10046}
2015-09-24 10:18:48 +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
Karl Wiberg
7404368998 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}
2015-09-22 17:31:52 +00:00
kwiberg
6faf5bebba Move AudioDecoderPcm* next to AudioEncoderPcm*
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1348613003

Cr-Commit-Position: refs/heads/master@{#10015}
2015-09-22 13:16:56 +00:00
kwiberg
ada4c130ab Move AudioDecoderG722 next to AudioEncoderG722
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1346993002

Cr-Commit-Position: refs/heads/master@{#9966}
2015-09-17 10:12:38 +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
Karl Wiberg
c0ac6cad00 Move AudioDecoderPcm16B next to AudioEncoderPcm16B
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1348113002 .

Cr-Commit-Position: refs/heads/master@{#9963}
2015-09-17 05:47:55 +00:00
kwiberg
fff9f176f5 Move AudioDecoderIlbc next to AudioEncoderIlbc
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557

Review URL: https://codereview.webrtc.org/1348053002

Cr-Commit-Position: refs/heads/master@{#9961}
2015-09-17 04:26:39 +00:00
Karl Wiberg
0b05879cd7 Move AudioDecoderOpus next to AudioEncoderOpus
All AudioDecoder subclasses have historically lived in NetEq, but they
fit better with the codec they wrap.

BUG=webrtc:4557
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1342933005 .

Cr-Commit-Position: refs/heads/master@{#9944}
2015-09-15 15:28:29 +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
kwiberg
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
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
Andrew MacDonald
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
Karl Wiberg
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