| 6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 4cf61dd116288e9f119209c59e07f1d9439d8d05 |
|
09-Dec-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4 |
|
23-Nov-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| ee2bac26dd3eb4463126098f87701ff66098b288 |
|
11-Nov-2015 |
kwiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| ee1879ca40ffe4af9bb9613e03eacc5c2c4881fc |
|
29-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 48ed930975ef7e84023044ed584c4eff914e6c9a |
|
29-Oct-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 74640895fafbb90a6630a6a91b80da0a7cff229c |
|
29-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
audio_coding: rename interface -> include
BUG=webrtc:5095
R=henrik.lundin@webrtc.org
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417173004 .
Cr-Commit-Position: refs/heads/master@{#10444}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 6d92bf59f3f8c0ce8ad445c11aaaf955eae752cc |
|
23-Sep-2015 |
minyuel <minyue@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 3c089d751ede283e21e186885eaf705c3257ccd2 |
|
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 9c3efd00523a81d0f2b582799fbe67afe44139b2 |
|
27-Aug-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 1bb8cf846d9a0bfe74fceae34ebef60f56d12fa4 |
|
25-Aug-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| dce40cf804019a9898b6ab8d8262466b697c56e0 |
|
24-Aug-2015 |
Peter Kasting <pkasting@google.com> |
Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 5abd3e1f986c627a852bf823d15feaa5f619a559 |
|
03-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| d8a03facf6986a011c8f889c63d87f9216a1e912 |
|
03-Jun-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| cf808d2366e58b33540931d182f36800d9a15b0d |
|
27-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| d8399e630f3f4886d455e2c4d2307794b60261c0 |
|
25-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 7f6c4d42a2605d1da39af3f957a46cf57b043b84 |
|
09-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: https://chromium.googlesource.com/chromium/src/+/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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| b0f4b3da055cb09813d52f417f64ce2275887fea |
|
04-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 7cbc4f969aa1f145b1538c0f0144ad3cc81b69e3 |
|
07-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 38344ed2806c8fed60d67d280ca44c32e36707c0 |
|
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 47658f12691138c464eddac1ad32cbc98b8de5a8 |
|
11-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| ea25784107b9202300a57d838d2c56e158220eef |
|
07-Aug-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 9c55f0f957534144d2b8a64154f0a479249b34be |
|
09-Jun-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| 1b9df05c8521d1d807b08d7c00eb2f7e5b097fdf |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
|
| a90f6d67f72359cf63b59480fa87a13aae808c03 |
|
28-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/audio_coding/neteq/neteq_impl.h
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