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/voice_engine/test/auto_test/standard/codec_test.cc
|
44307630d3ad6dcdd7b7fd07e78881b50a92ced4 |
|
16-Dec-2015 |
kwiberg <kwiberg@webrtc.org> |
AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders All encoders already handle the "Opus-specific" requests sanely (by failing nicely), so we don't need extra checks to protect them. BUG=webrtc:5028 Review URL: https://codereview.webrtc.org/1527453005 Cr-Commit-Position: refs/heads/master@{#11051}
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
5c389d3e09646c0e2ed76d5ccb37a3419a09eb6a |
|
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Split webrtc/video into webrtc/{audio,call,video}. Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts into webrtc/call, splitting out audio/shared components with separate OWNERS files. BUG=webrtc:4690 R=solenberg@webrtc.org, tina.legrand@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1227923005 . Cr-Commit-Position: refs/heads/master@{#10073}
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
b04965ccf83c2bc6e2758abab9bea0c18551a54c |
|
09-Sep-2015 |
ivoc <ivoc@webrtc.org> |
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. An option was added to voe_cmd_test to make a RtcEventLog dump. BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1267683002 Cr-Commit-Position: refs/heads/master@{#9901}
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
9b2e1144df6e3622354caca00baf4a7462a0809c |
|
13-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Supporting Opus DTX in Voice Engine. Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API. BUG=1014 R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43709004 Cr-Commit-Position: refs/heads/master@{#8716} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
664ccb7d8da3adfffdb7c56f885b633224555e6e |
|
28-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland r8125: Modify some tests to never use DTX disable mode DTX disable mode will be removed as a part of the ACM redesign work. This CL effectively reverts r8129, and relands r8125, but now using assert instead of DCHECK. COAUTHOR:kwiberg@webrtc.org TBR=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37839004 Cr-Commit-Position: refs/heads/master@{#8185} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8185 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
ff108fe5084e8326e7c38be728557758bd58b1e7 |
|
22-Jan-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Revert 8125 "Modify some tests to never use DTX disable mode" Broke compile on the Chromium FYI bots: http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483 http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028 http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293 Error: In file included from ../../third_party/webrtc/voice_engine/channel.cc:13: In file included from ../../third_party/webrtc/base/checks.h:22: In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35: ../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined] #define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity)) ^ ../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here #define LOG(sev) \ ^ In file included from ../../third_party/webrtc/voice_engine/channel.cc:13: In file included from ../../third_party/webrtc/base/checks.h:22: ../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined] #define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0) ^ ../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here #define LOG_V(sev) \ ^ 2 errors generated. > Modify some tests to never use DTX disable mode > > DTX disable mode will be removed as a part of the ACM redesign work. > > COAUTHOR:kwiberg@webrtc.org > > R=henrika@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/34769004 TBR=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35859004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
043db247676ebec3b4a166a004c7cce7687962fc |
|
22-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Modify some tests to never use DTX disable mode DTX disable mode will be removed as a part of the ACM redesign work. COAUTHOR:kwiberg@webrtc.org R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
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/voice_engine/test/auto_test/standard/codec_test.cc
|
6aac93bd9c3da92e92b016d83c8f84c65aae65b6 |
|
12-Aug-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding SetOpusMaxBandwidth in VoE and ACM This is a step to solve https://code.google.com/p/webrtc/issues/detail?id=1906 In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth. TEST = added a test in voe_cmd_test and listened to the result BUG= R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
0f7375504a98e43101f682143ae8f3866aec3ed3 |
|
17-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. BUG=3206 R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in voice_engine/ BUG=1662 R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1434005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
8a025e26db985a128023f43dfa6abc7b9c2d1a72 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure VoiceEngine tests only include one test framework. BUG= R=henrikg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1499004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4078 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
|
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/voice_engine/test/auto_test/standard/codec_test.cc
|