| 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/packet_buffer.h
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| dce40cf804019a9898b6ab8d8262466b697c56e0 |
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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/packet_buffer.h
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| 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/packet_buffer.h
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| 52b42cb069a035f10e951195c28cf6d05d1fd91c |
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04-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Fix problem with late packets in NetEq
Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.
BUG=chrome:423985
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
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| 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/packet_buffer.h
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| 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/packet_buffer.h
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| 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/packet_buffer.h
|
| c3e8abda7c97f568c9c8fc06b6d48b1fe4b65c33 |
|
13-May-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Deleting all NetEq3 files
NetEq3 is deprecated and replaced by NetEq4
(webrtc/modules/audio_coding/neteq4/).
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14469007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6118 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|
| 28d54ab18f18c7730105282d83df8a50a6b13e89 |
|
22-Apr-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improve AV-sync when initial delay is set and NetEq has long buffer.
Review URL: https://webrtc-codereview.appspot.com/1324006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3883 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|
| 0946a56023d821e0deca04029bb016ae1f23aa82 |
|
09-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t etc. in audio_coding/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1271006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|
| 6388c3e2fdfc91b3648fb7d408a14ddb25e41cd1 |
|
12-Feb-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.
TEST=ACM unit test is added, also a manual integration test is writen.
Review URL: https://webrtc-codereview.appspot.com/1097009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|
| 5ac387c4d1434fc086f3f159bdde068a5994feed |
|
19-Nov-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Allow NetEQ to use real packet durations.
This is a copy of http://review.webrtc.org/864014/
This adds a FuncDurationEst to each codec instance which estimates
the duration of a packet given the packet contents and the duration
of the previous packet. By default, this simply returns the
duration of the previous packet (which is what is currently assumed
to be the duration of all future packets). This patch also provides
an initial implementation of this function for G.711 which returns
the actual number of samples in the packet.
BUG=issue1015
Review URL: https://webrtc-codereview.appspot.com/935016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/packet_buffer.h
|