| 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/codecs/opus/opus_interface.c
|
| 3c652b67468d182bd36aee4c31557621be50cc92 |
|
18-Nov-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 3cea25680620f0b7df7642fc8fe49d0ecaf8b466 |
|
10-Nov-2015 |
minyue <minyue@webrtc.org> |
Reland "Prevent Opus DTX from generating intermittent noise during silence"
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/
A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.
BUG=
Review URL: https://codereview.webrtc.org/1422213003
Cr-Commit-Position: refs/heads/master@{#10574}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| b4a753fdb5725e1b241c6c40cc2a752e57cfbdcb |
|
09-Nov-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
Reason for revert:
Breaks voe_auto_test on all three "large tests bots".
https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio
Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages).
Original issue's description:
> Prevent Opus DTX from generating intermittent noise during silence.
>
> Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
>
> BUG=webrtc:5127
>
> Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977
> Cr-Commit-Position: refs/heads/master@{#10565}
TBR=tina.legrand@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1428613004
Cr-Commit-Position: refs/heads/master@{#10567}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| f475add57eada116bc960fe2935876ec8c077977 |
|
09-Nov-2015 |
minyue <minyue@webrtc.org> |
Prevent Opus DTX from generating intermittent noise during silence.
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1415173005
Cr-Commit-Position: refs/heads/master@{#10565}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 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/codecs/opus/opus_interface.c
|
| 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/codecs/opus/opus_interface.c
|
| 4376648df021fd82f25a38694e33678f802d06ea |
|
27-Aug-2015 |
Karl Wiberg <kwiberg@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 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/codecs/opus/opus_interface.c
|
| bba780707848ad36066739c60d7b28cd752fb92f |
|
12-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| cb180976dd0e9672cde4523d87b5f4857478b5e9 |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 83ad33a8aed1fb00e422b6abd33c3e8942821c24 |
|
10-Jun-2015 |
Peter Kasting <pkasting@google.com> |
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}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 092041c1cdadeb82463ee79dfc291d60b41d35ef |
|
11-May-2015 |
Minyue Li <minyue@webrtc.org> |
Setting OPUS_SIGNAL_VOICE when enable DTX.
A better solution than forcing OPUS_APPLICATION_VOIP when enabling DTX has been found, which is to set OPUS_SIGNAL_VOICE.
This reduces the uncertainty of entering DTX over silence period of audio.
This CL contains the setup of OPUS_SIGNAL_VOICE and decoupling opus application mode with DTX.
BUG=4559
R=henrik.lundin@webrtc.org, henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46959004
Cr-Commit-Position: refs/heads/master@{#9168}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 7dba7860c79652593f0a643fc81fe35f8707e0db |
|
20-Jan-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Setting Opus target application.
This CL is to allow to set Opus target application at the creation of an encoder.
According to Opus spec, there are three applications:
OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY
BUG=
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 0ca768b13197d2c1e7411ccbb9a693e1f7eaad0a |
|
11-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding DTX to WebRTC Opus wrapper (relanding).
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.
See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/
Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 19dd129c69956ac8a7fb6150cd15694f720cad19 |
|
09-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Revert 7846 "Adding DTX to WebRTC Opus wrapper"
> Adding DTX to WebRTC Opus wrapper
>
> This is a step toward adding Opus DTX support in WebRTC.
>
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
>
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
>
> We transmit the first 1-byte packet to let decoder be in-sync
>
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13219004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 4321f175f1d2e6cfe1e56ece176c258f17101e83 |
|
09-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding DTX to WebRTC Opus wrapper
This is a step toward adding Opus DTX support in WebRTC.
Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
We transmit the first 1-byte packet to let decoder be in-sync
BUG=webrtc:1014
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 33ccdfa1f555e00170e2b98cd0f575eed3e46236 |
|
04-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Relanding r7807.
r7807 was reverted to be excluded from the cause of a failure.
It has been verified and can reland now.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 52bc4f47973b68bf78b9587bf4856e9bbf5784ed |
|
04-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Revert 7807 "Removing unused opus wrapper APIs."
> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| e54a6342dd52f95b0d7647daeb984cb94ac88263 |
|
04-Dec-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Removing unused opus wrapper APIs.
WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
WebRtcOpus_DecodePlcMaster/Slave() are also removed.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 4bd2db9a556a7a889daf3812bc9e092f5f3cf536 |
|
09-Oct-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Opus wrapper: Use const for inputs and uint8[] for byte streams
About half of the functions already followed the desired pattern; this
patch fixes the other half.
BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7409 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| adee8f924224e116f041564ddde83c979880e35f |
|
03-Sep-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 0040a6ef97236053d9698470b9d4c095e8019f1c |
|
04-Aug-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This is a setup to solve
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).
BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| f563e85ab0bac7d2f5e70f70af7790595726832b |
|
18-Jul-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This is to re-open an earlier CL
https://webrtc-codereview.appspot.com/16619005/
which is reverted due to an issue in audio conference mixer.
This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/
BUG=webrtc:3155
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18819005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| d42da54768cfb8319c38e5403ce147193dbe1095 |
|
17-Jun-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
>
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/16619005
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 8f8503d947e820cce35fa3d0f2b25b6b893cf141 |
|
17-Jun-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
TEST=passed_all_trybots
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
|
| 46509c8d582404d224d484fcf28262b610a5fbec |
|
07-Mar-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
adding FEC support to WebRTC Opus wrapper and tests.
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 04546884bf7f816e52e1a6db03d6bba49a12edc5 |
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07-Mar-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
This CL is to add Opus complexity knob and to test it.
As a by-product, a generic tool for testing and comparing the complexity of codecs is added, and new audio files are added to the resources.
Three complexity tests are included
1. Default Opus complexity
2. Opus complexity knob
3. Default iSAC complexity (to compare with Opus)
The complexity tests are only meant for development reasons
and not to be run at bots.
The .isolate file is only needed for the APK packaging and test execution on Android.
TEST=passes all trybots
BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5655 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| bd21fb5f8dbe5345737972475782f693e698f541 |
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08-Aug-2013 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding call to Opus PLC
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.
BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1727004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 45426eadf564727cc25e304ad8cebae25a21f0af |
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03-Jul-2013 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
In call to Opus decoder: frame length too large
BUG=https://code.google.com/p/webrtc/issues/detail?id=1201
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1752004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4292 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| db11fab49efc974cfd645fe16f345b9cb3eba91b |
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17-Apr-2013 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding Opus unit test
This CL adds a unit test for Opus, as well as new APIs for true stereo decoding (skipping master/slave approach).
BUG=
Review URL: https://webrtc-codereview.appspot.com/1222006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3860 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 46d90dcd740d0f63e52ece2dc1a1d27c56e222a1 |
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01-Feb-2013 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding three frame sizes to Opus
Adding support for 10, 40 and 60 ms packet sizes for Opus.
BUG=issue1015
Review URL: https://webrtc-codereview.appspot.com/1086004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3454 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 5dfb1f2cd31c07708c32947375718245ef280724 |
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23-Jan-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Bug fix in WebRtcOpus_DurationEst
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.
BUG=1334
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| d0d41498a32c6723e74b7778d9600ec230fa4eb2 |
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20-Dec-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding AUDIO application as default for Opus stereo
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.
I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.
Next step is to add an API to choose application mode.
BUG=issue1239
Review URL: https://webrtc-codereview.appspot.com/1007006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 4275ab1ca03157807bc26d7429a57e349d65f6b3 |
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19-Dec-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement NetEq duration estimation for Opus.
Review URL: https://webrtc-codereview.appspot.com/983004
Patch from Ralph Giles <giles@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3314 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| c4590580e8de143443b88fde53518b5ed8c9ce76 |
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28-Nov-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Opus mono/stereo on the same payloadtype, and fix of memory bug
During call setup Opus should always be signaled as a 48000 Hz stereo codec, not depending on what we plan to send, or how we plan to decode received packets.
The previous implementation had different payload types for mono and stereo, which breaks the proposed standard.
While working on this CL I ran in to the problem reported earlier, that we could get a crash related to deleting decoder memory. This should now be solved in Patch Set 3.
BUG=issue1013, issue1112
Review URL: https://webrtc-codereview.appspot.com/933022
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3177 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 0ad3c1af0a66ec6fb54b0cef0ed3c42fa407157f |
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07-Nov-2012 |
tina.legrand@webrtc.org <tina.legrand@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding Opus stereo support to WebRTC
This CL adds support for sending and receiving stereo using the Opus codec.
BUG=issue1013
Review URL: https://webrtc-codereview.appspot.com/930008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3050 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
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| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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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/codecs/opus/opus_interface.c
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