| 92f8dbde77f859449a2b9ac107bca6c9b4329897 |
|
24-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VIDEOCODEC_* from engine_configurations.h.
Removes index-based codec fetching from the VCM and overall cleans up
the code.
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1425613004 .
Cr-Commit-Position: refs/heads/master@{#10770}
/external/webrtc/webrtc/engine_configurations.h
|
| 98ab3a46d6b98bd6626ab741092f7cbf104d127b |
|
01-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/engine_configurations.h
|
| 3fd7be4cb1d41ff6298a90c17acf52d379ab8812 |
|
25-Sep-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/engine_configurations.h
|
| f66a9251424351ea6d631c54dd1feb64cc13d809 |
|
24-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
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}
/external/webrtc/webrtc/engine_configurations.h
|
| e510d7f100f716048a216e2786617d1bbd5bb815 |
|
18-Sep-2015 |
henrik.lundin <henrik.lundin@webrtc.org> |
Remove ACM AudioCodingFeedback callback object and derived classes
The callback object was not used anymore. Also removing the deprecated
WEBRTC_DTMF_DETECTION macro from engine_configurations.h.
BUG=3520
Review URL: https://codereview.webrtc.org/1353763002
Cr-Commit-Position: refs/heads/master@{#9988}
/external/webrtc/webrtc/engine_configurations.h
|
| 1f9baab753be55a7c6d31c84a5470fe646936edd |
|
17-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Remove the preprocessor symbol WEBRTC_CODEC_AVT (it was always defined)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1338283002
Cr-Commit-Position: refs/heads/master@{#9960}
/external/webrtc/webrtc/engine_configurations.h
|
| 844a91081ef1141bd9888e828bef87a7737c24a8 |
|
16-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Remove the preprocessor symbol WEBRTC_CODEC_PCM16 (it was always defined)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1336923002
Cr-Commit-Position: refs/heads/master@{#9955}
/external/webrtc/webrtc/engine_configurations.h
|
| 300eeb68f55c5091c7045e377578586733cddf16 |
|
12-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interfaces.
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.
Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).
BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55379004
Cr-Commit-Position: refs/heads/master@{#9179}
/external/webrtc/webrtc/engine_configurations.h
|
| fa5874544577cb1b59657e7c79d9eb51383c855d |
|
23-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Delete all codec-specific subclasses of ACMGenericCodec
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)
This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc in r8244.
BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40729004
Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 5b8831782074d490969171de5f8c67251f36d9cc |
|
01-Nov-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932
R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| cfe3845b668d20b917d22d15ee3dd1b1e668d465 |
|
29-Oct-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Enable G.722 for Chromium builds
BUG=3909
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7555 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| b1dac33cac5a64cbec6b0fd72624fa9d3060376c |
|
17-Oct-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
BUG=3932
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/27779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 573c78e31c7ccdc5cf44ebc54b9fc089f5e8f0cf |
|
10-Oct-2014 |
marpan@webrtc.org <marpan@webrtc.org> |
Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.
R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| b9f5453e2997253addb87706a43b4484e1139972 |
|
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add boilerplate code for H.264.
R=mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 1cec3957b88cbab345535137329bd8f3f2a6b39e |
|
12-May-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| b9309beea40e1fd99297d4658a16864a801329c3 |
|
14-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
BUG=3206
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| a07923339bea76571f2f9ac33316eb56dfb47054 |
|
18-Feb-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove external encryption API for VoE.
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| fc320466d12e16c1e80f57b8cff864627f2766f6 |
|
11-Feb-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove ViE external encryption API.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5525 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| bf00740c92839865f3656fb4ee02b144f26b2012 |
|
17-Sep-2013 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a new voice engine warning for the typing noise off state.
The old VE_TYPING_NOISE_WARNING is unchanged and fired whenever typing noise is detected.
The new VE_TYPING_NOISE_OFF_WARNING is fired when typing noise was detected and is gone now.
This is necessary for converting the typing state to a PeerConnection stats.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4770 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 8fa03a15ab5fa7fd600888d20363736b00387dfb |
|
12-Sep-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make PCM16 available in Chromium builds.
PCM16 can be useful for unit tests in Chromium. In particular Mikhal
would like to use it for ChromeCast.
This currently (r222592) has no impact on Chrome binary size, presumably
because PCM16 is unused and the linker strips the symbols.
To measure the potential impact, I looked at the size (bytes) of
out/Release/vie_auto_test on Linux with various codecs removed:
r4724 : 4567384
No PCM16 : 4565936
No ILBC : 4500424
No G722 : 4555800
No RED : 4565880
Giving the following size increases of adding each codec:
PCM16 : 1.4 kB (0.03%)
ILBC : 70.0 kB (1.49%)
G722 : 11.6 kB (0.25%)
RED : 1.5 kB (0.03%)
R=mikhal@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2195005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4732 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 0851df8d60b43e1c7a212f233dc378cb2585476b |
|
19-Jun-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.
R=niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1683004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| b9e402d99f25d879fd62777e6646e734be07348b |
|
04-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove WEBRTC_*_ENGINE_NETWORK_API use
Review URL: https://webrtc-codereview.appspot.com/1203009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| a442d4d98337bc25e4c469e20fde62aab33e2f59 |
|
28-Mar-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
Today I had to figure out this code was legacy. Now next person doesn't have to.
BUG=
Review URL: https://webrtc-codereview.appspot.com/1247004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 684f0577fbe4ea393fef1dddf2ca7d02e3205b49 |
|
14-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 361bac7a4f30a81e58c53ba86c58ffec085306d7 |
|
13-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.h
|
| 6bd737a714ee2f67124aafd2b40ac3b36ff08ef8 |
|
04-Dec-2012 |
dwkang@webrtc.org <dwkang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
First pass of MediaCodecDecoder which uses Android MediaCodec API.
Background:
As of now, MediaCodec API is the only public interface which enables us
to access low level HW resource in Android. ViEMediaCodecDecoder will be
used for further experiments/exploration.
TODO:
To fix known issues. (detaching thread from VM and frequent GC)
Review URL: https://webrtc-codereview.appspot.com/933033
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3233 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/engine_configurations.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/engine_configurations.h
|