47 Commits

Author SHA1 Message Date
henrik.lundin
da8bbf6e3c NetEq: Change member variables for current RTP types to rtc::Optionals
With this change, the value 0xFF is no longer used to flag that the RTP
type is unknown. Instead, an empty value for the rtc::Optional is used.

Review-Url: https://codereview.webrtc.org/2290153002
Cr-Commit-Position: refs/heads/master@{#13989}
2016-08-31 10:14:18 +00:00
henrik.lundin
b3f1c5d2fe Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine
The new method returns the current total delay (packet buffer and sync
buffer) in ms, with smoothing applied to even out short-time
fluctuations due to jitter. The packet buffer part of the delay is not
updated during DTX/CNG periods.

This CL also pipes the new metric through ACM and uses it in
VoiceEngine. It replaces the previous method of estimating the buffer
delay (where an inserted packet's RTP timestamp was compared with the
last played timestamp from NetEq). The new method works better under
periods of DTX/CNG.

Review-Url: https://codereview.webrtc.org/2262203002
Cr-Commit-Position: refs/heads/master@{#13855}
2016-08-22 22:40:00 +00:00
kwiberg
342f74005f NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument
BUG=webrtc:5801
NOTRY=true

Review-Url: https://codereview.webrtc.org/2027993002
Cr-Commit-Position: refs/heads/master@{#13162}
2016-06-16 10:18:09 +00:00
henrik.lundin
919518613f NetEq: Rename Nack to NackTracker to avoid name collisions in GN
BUG=webrtc:5949
NOTRY=True
R=kjellander@webrtc.org

Review-Url: https://codereview.webrtc.org/2045243002
Cr-Commit-Position: refs/heads/master@{#13069}
2016-06-08 13:43:49 +00:00
ossu
e352578bc8 Moved injection of AudioDecoderFactory into voe::Channel.
Channel's API remains unchanged, but the creation of a BuiltinAudioDecoderFactory is now in Channel. The next step would be to amend Channel's API (through CreateChannel, I believe) to allow an AudioDecoderFactory to be sent along.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1992763002
Cr-Commit-Position: refs/heads/master@{#12893}
2016-05-25 14:37:47 +00:00
henrik.lundin
7a926812d8 NetEq: Implement muted output
This CL implements the muted output functionality in NetEq. Tests are
added. The feature is currently off by default, and AcmReceiver makes
sure that the muted state is not engaged.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1965733002
Cr-Commit-Position: refs/heads/master@{#12711}
2016-05-12 20:51:37 +00:00
minyue
1746179c96 Reducing neteq sync buffer size.
BUG=608644

Review-Url: https://codereview.webrtc.org/1947453002
Cr-Commit-Position: refs/heads/master@{#12614}
2016-05-03 20:32:13 +00:00
henrik.lundin
b1fb72bebb NetEq: Move counting of generated CNG samples from DecisionLogic
The counting is moved to NetEqImpl, and the new counter is realized as a
Stopwatch object. The DecisionLogic class still has to maintain record
of when the CNG period is shortened, in order to reduce the delay. This
is recorded in a new noise_fast_forward_ member in DecisionLogic.

BUG=webrtc:5608

Review-Url: https://codereview.webrtc.org/1914303004
Cr-Commit-Position: refs/heads/master@{#12608}
2016-05-03 15:18:54 +00:00
minyue
5bd3397e53 Adding 120 ms frame length support in NetEq.
BUG=webrtc:1015

Review-Url: https://codereview.webrtc.org/1901633002
Cr-Commit-Position: refs/heads/master@{#12592}
2016-05-02 11:46:19 +00:00
henrik.lundin
1d9061e8ae NetEq: Dependency injection through one struct instead of many params
With this change, the NetEqImpl constructor takes a struct
(NetEqImpl::Dependencies) as input instead of a collection of
individual dependencies. The NetEqImpl unit test fixture is modified
to make better used of unique_ptrs.

Review URL: https://codereview.webrtc.org/1921243002

Cr-Commit-Position: refs/heads/master@{#12514}
2016-04-26 19:19:43 +00:00
henrik.lundin
ed4972116c Add a TickTimer object to NetEqImpl
The TickTimer is incremented on each call to GetAudioInternal(). Other
than that, the new object is not used yet.

Also adding a unit test in NetEqImplTest to verify that the tick timer
is incremented in the call to NetEq::GetAudio.

BUG=webrtc:5608

Review URL: https://codereview.webrtc.org/1903153005

Cr-Commit-Position: refs/heads/master@{#12493}
2016-04-25 17:11:45 +00:00
henrik.lundin
15c51e355f Move setting of AudioFrame::timestamp_ into NetEq
This was previously done in AcmReceiver, but belongs in NetEq where the
rest of the AudioFrame fields are populated.

BUG=webrtc:5669,webrtc:5607

Review URL: https://codereview.webrtc.org/1863993002

Cr-Commit-Position: refs/heads/master@{#12265}
2016-04-06 15:39:01 +00:00
henrik.lundin
9a410dd082 Change NetEq::GetPlayoutTimestamp to return an rtc::Optional<uint32_t>
This is in preparation for changes to when the playout timestamp is
valid.

BUG=webrtc:5669

Review URL: https://codereview.webrtc.org/1853183002

Cr-Commit-Position: refs/heads/master@{#12256}
2016-04-06 08:39:30 +00:00
henrik.lundin
55480f5efa Remove the type parameter to NetEq::GetAudio
The type is included in the AudioFrame output parameter.

Rename the type NetEqOutputType to just OutputType, since it is now
internal to NetEq.

BUG=webrtc:5607

Review URL: https://codereview.webrtc.org/1769883002

Cr-Commit-Position: refs/heads/master@{#11903}
2016-03-08 10:38:02 +00:00
henrik.lundin
500c04bc86 Delete VAD methods from AcmReceiver and move functionality inside NetEq
This change essentially does two things:

1. Remove the VAD-related methods from AcmReceiver. These are
EnableVad(), DisableVad(), and vad_enabled(). None of them were used
outside of unit tests.

2. Move the functionality to set AudioFrame::speech_type_ and
AudioFrame::vad_activity_ inside NetEq. This was previously done in
AcmReceiver, but based on information inherently owned by NetEq.

With the change in 2, NetEq's GetAudio interface can be simplified by
removing the output type parameter. This will be done in a follow-up
CL.

BUG=webrtc:5607

Review URL: https://codereview.webrtc.org/1772583002

Cr-Commit-Position: refs/heads/master@{#11902}
2016-03-08 10:36:07 +00:00
henrik.lundin
6d8e011b64 Change NetEq::GetAudio to use AudioFrame
With this change, NetEq now uses AudioFrame as output type, like the
surrounding functions in ACM and VoiceEngine already do.

The computational savings is probably slim, since one memcpy is
removed while another one is added (both in AcmReceiver::GetAudio).

More simplifications and clean-up will be done in
AcmReceiver::GetAudio in future CLs.

BUG=webrtc:5607

Review URL: https://codereview.webrtc.org/1750353002

Cr-Commit-Position: refs/heads/master@{#11874}
2016-03-04 18:34:26 +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
pbos
5ad935cb56 Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1613643004

Cr-Commit-Position: refs/heads/master@{#11367}
2016-01-25 11:52:53 +00:00
Tommi
9090e0b147 Switch CriticalSectionWrapper->rtc::CriticalSection in modules/audio_coding.
This is a part of cleaning up CriticalSectionWrapper in general.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1610073003 .

Cr-Commit-Position: refs/heads/master@{#11319}
2016-01-20 12:39:45 +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
henrik.lundin
4cf61dd116 NetEq: Add codec name and RTP timestamp rate to DecoderInfo
The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1484343003

Cr-Commit-Position: refs/heads/master@{#10952}
2015-12-09 14:21:02 +00:00
henrik.lundin
d89814bfd7 NetEq: Add new method last_output_sample_rate_hz
This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1467163002

Cr-Commit-Position: refs/heads/master@{#10754}
2015-11-23 14:49:31 +00:00
kwiberg
ee2bac26dd AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments
Instead of separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1429943004

Cr-Commit-Position: refs/heads/master@{#10606}
2015-11-11 18:34:07 +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.lundin
48ed930975 ACM: Move NACK functionality inside NetEq
Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.

This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
  forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1410073006

Cr-Commit-Position: refs/heads/master@{#10448}
2015-10-29 12:36:32 +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
minyuel
6d92bf59f3 Returning correct duration estimate on Opus DTX packets.
Bug 4985 revealed two flaws
1. Opus duration estimate did not return correct length for DTX packets,

2. NetEq DoCodecInternalCng did not assign enough buffer.

P.S.
Generalizing problem 1, current NetEq decode function checks memory size by calling the duration estimate function. This is not ideal. A better way is to let codec's decode function to receive buffer size and return failure if it is not enough. This can be made in a separate CL.

BUG=webrtc:4985
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1334303005 .

Cr-Commit-Position: refs/heads/master@{#10031}
2015-09-23 13:20:56 +00:00
henrikg
3c089d751e Add RTC_ prefix to contructormagic 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.

* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS

Related CL: https://codereview.webrtc.org/1335923002/

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1345433002

Cr-Commit-Position: refs/heads/master@{#9953}
2015-09-16 12:37:52 +00:00
henrik.lundin
9c3efd0052 Reland: Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay
(packet buffer and sync buffer) of NetEq. This is the same information
that was already available in
NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained
through NetEq::NetworkStatistics(). But, since the current delay is a
key metric of NetEq, it is convenient to have it available in a
simpler way.

This is a re-landing of r9359,
https://webrtc-codereview.appspot.com/51149004, which was reverted in
r9360. The refactoring made in r9669 facilitated the relanding.

TBR=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1313873003

Cr-Commit-Position: refs/heads/master@{#9801}
2015-08-27 20:12:27 +00:00
Henrik Lundin
1bb8cf846d NetEq/ACM: Refactor how packet waiting times are calculated
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.

R=ivoc@webrtc.org, minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1296633002 .

Cr-Commit-Position: refs/heads/master@{#9778}
2015-08-25 11:08:17 +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
5abd3e1f98 Revert r9359 "Implement NetEq's CurrentDelay function"
This reverts commit d8a03facf6986a011c8f889c63d87f9216a1e912, since it
broke the Chrome build. Will have to swap to using base/logging.h in
neteq_impl.cc before re-landing this change.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50219004

Cr-Commit-Position: refs/heads/master@{#9360}
2015-06-03 10:58:52 +00:00
Henrik Lundin
d8a03facf6 Implement NetEq's CurrentDelay function
This was not implemented before. It returns the current total delay (packet buffer and sync buffer) of NetEq. This is the same information that was already available in NetEqNetworkStatistics::current_buffer_size_ms, that can be obtained through NetEq::NetworkStatistics(). But, since the current delay is a key metric of NetEq, it is convenient to have it available in a simpler way.

R=kwiberg@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51149004

Cr-Commit-Position: refs/heads/master@{#9359}
2015-06-03 09:55:53 +00:00
Henrik Lundin
cf808d2366 Add new fast mode for NetEq's Accelerate operation
This change instroduces a mode where the Accelerate operation will be
more aggressive. When enabled, it will allow acceleration at lower
correlation levels, and possibly remove multiple pitch periods at
once.

The feature is enabled through NetEq::Config, and is off by
default. This means that bit-exactness tests are currently not
affected.

A unit test was added for the Accelerate class, with and without fast
mode enabled.

BUG=4691
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50039004

Cr-Commit-Position: refs/heads/master@{#9295}
2015-05-27 12:33:39 +00:00
Karl Wiberg
d8399e630f Also provide sample rate when registering decoders
This replaces the old practice of looking up the sample rate in a
table, which won't work when we add support for external decoders.

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

Review URL: https://webrtc-codereview.appspot.com/54469004

Cr-Commit-Position: refs/heads/master@{#9276}
2015-05-25 12:40:05 +00:00
Karl Wiberg
7f6c4d42a2 Fix clang style warnings in webrtc/modules/audio_coding/neteq
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.

BUG=163
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44109004

Cr-Commit-Position: refs/heads/master@{#8960}
2015-04-09 13:44:23 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
kwiberg@webrtc.org
00b8f6b364 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
2015-02-26 14:43:50 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
henrik.lundin@webrtc.org
b0f4b3da05 Improving error message from neteq_rtpplay
If a packet with unknown RTP payload type is inserted, this CL
will make sure that the error message is a little more detailed
and gives a better understadning of what to do.

BUG=2692
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7603 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 08:53:10 +00:00
henrik.lundin@webrtc.org
7cbc4f969a Set NetEq playout mode through the Config struct
This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 06:37:39 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 06:05:00 +00:00
henrike@webrtc.org
47658f1269 Mark all virtual overrides in the hierarchy of AudioPacketizationCallback,
RTPStream, and NetEq as such.  Also mark all other virtual overrides in the same
files.

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

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

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

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7144 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 22:14:59 +00:00
henrik.lundin@webrtc.org
ea25784107 Change how background noise mode in NetEq is set
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.

In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 12:27:37 +00:00
henrik.lundin@webrtc.org
9c55f0f957 Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.

This CL effectively reverts r6257 "Rename neteq4 folder to neteq".

BUG=2996
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-09 08:10:28 +00:00
henrik.lundin@webrtc.org
1b9df05c85 Revert 6257 "Rename neteq4 folder to neteq"
> Rename neteq4 folder to neteq
> 
> BUG=2996
> R=turaj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12569005

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6259 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 07:33:39 +00:00
henrik.lundin@webrtc.org
a90f6d67f7 Rename neteq4 folder to neteq
BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6257 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-28 06:23:34 +00:00