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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
288886b2ec9a2dac730f115e9c3079d8439efe60 |
|
06-Nov-2015 |
kwiberg <kwiberg@webrtc.org> |
Pass audio to AudioEncoder::Encode() in an ArrayView Instead of in separate pointer and size arguments. Review URL: https://codereview.webrtc.org/1418423010 Cr-Commit-Position: refs/heads/master@{#10535}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
1d34fe979c52e5826c5c8162759b0167b2607836 |
|
16-Jun-2015 |
henrika <henrika@chromium.org> |
Adds support for webrtc::test::ResourcePath on iOS BUG=webrtc:4752 R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1178843002. Cr-Commit-Position: refs/heads/master@{#9445}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
728d9037c016c01295177fa700fc7927f0bb80bb |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Reformat existing code. There should be no functional effects. This includes changes like: * Attempt to break lines at better positions * Use "override" in more places, don't use "virtual" with it * Use {} where the body is more than one line * Make declaration and definition arg names match * Eliminate unused code * EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT) * Correct #include order * Use anonymous namespaces in preference to "static" for file-scoping * Eliminate unnecessary casts * Update reference code in comments of ARM assembly sources to match actual current C code * Fix indenting to be more style-guide compliant * Use arraysize() in more places * Use bool instead of int for "boolean" values (0/1) * Shorten and simplify code * Spaces around operators * 80 column limit * Use const more consistently * Space goes after '*' in type name, not before * Remove unnecessary return values * Use "(var == const)", not "(const == var)" * Spelling * Prefer true, typed constants to "enum hack" constants * Avoid "virtual" on non-overridden functions * ASSERT(x == y) -> ASSERT_EQ(y, x) BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://codereview.webrtc.org/1172163004 Cr-Commit-Position: refs/heads/master@{#9420}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
f045e4da43e671ae511aa1d9b6ef2968256a745d |
|
11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Prepare to convert various types to size_t. This makes some behaviorally-invariant changes to make certain code that currently only works correctly with signed types work safely regardless of the signedness of the types in question. This is preparation for a future change that will convert a variety of types to size_t. There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants. BUG=none R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=ajm Review URL: https://codereview.webrtc.org/1174813003 Cr-Commit-Position: refs/heads/master@{#9413}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
a2c79405b407162119954d57855c8c04c043df76 |
|
10-Jun-2015 |
henrika <henrika@chromium.org> |
Ensures that modules_unittests runs on iOS BUG=4752 R=tkchin@chromium.org Review URL: https://codereview.webrtc.org/1171033002. Cr-Commit-Position: refs/heads/master@{#9408}
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
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_unittest.cc
|
7f7d7e3427cc70e1b8b050283ef031e28c83699a |
|
16-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Prevent crash in NetEQ when decoder overflow. NetEQ can crash when decoder gives too many output samples than it can handle. A practical case this happens is when multiple opus packets are combined. The best solution is to pass the max size to the ACM decode function and let it return a failure if the max size if too small. BUG=4361 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45619004 Cr-Commit-Position: refs/heads/master@{#8730} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8730 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
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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_unittest.cc
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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_unittest.cc
|
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_unittest.cc
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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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
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_unittest.cc
|
46509c8d582404d224d484fcf28262b610a5fbec |
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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_unittest.cc
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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_unittest.cc
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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_unittest.cc
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3004c79c6ad0ca4b4df27d0ca76c2eb29735e267 |
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07-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix clang errors in non-GYP_DEFINES=clang=1 build BUG=1623 R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1368004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
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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_unittest.cc
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