d7b7ae8bdaaf51d885bbbcf6d2ccd2da2522ab03 |
|
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Add encode/decode time tracing to audio_coding. Also removes virtual from VideoDecoder::Decode and updated mocks and tests accordingly to use VideoDecoder::DecodeInternal instead. BUG=webrtc:5167 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1512483003 . Cr-Commit-Position: refs/heads/master@{#10935}
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
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/neteq/mock/mock_external_decoder_pcm16b.h
|
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/mock/mock_external_decoder_pcm16b.h
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4376648df021fd82f25a38694e33678f802d06ea |
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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/neteq/mock/mock_external_decoder_pcm16b.h
|
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/mock/mock_external_decoder_pcm16b.h
|
6dba1ebd14d8cd96e6e56adad868b33fdedecc53 |
|
18-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Make AudioDecoder stateless The channels_ member varable is removed from the base class, and the associated accessor function is changed to Channels() which is a pure virtual function. R=jmarusic@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43779004 Cr-Commit-Position: refs/heads/master@{#8775} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8775 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
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/neteq/mock/mock_external_decoder_pcm16b.h
|
1eda4e3db60f484d179cee359e150c4f0c9c7c67 |
|
25-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Reland r8476 "Set decoder output frequency in AudioDecoder::Decode call" This should be safe to land now that issue 4143 was resolved (in r8492). This change effectively reverts 8488. TBR=kwiberg@webrtc.org Original commit message: This CL changes the way the decoder sample rate is set and updated. In practice, it only concerns the iSAC (float) codec. One single iSAC decoder instance is used for both wideband and super-wideband decoding, and the instance must be told to switch output frequency if the payload type changes. This used to be done through a call to UpdateDecoderSampleRate, but is now instead done in the Decode call as an extra parameter. Review URL: https://webrtc-codereview.appspot.com/39289004 Cr-Commit-Position: refs/heads/master@{#8496} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8496 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
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903182bd8e782b162900b99bc7e25c35edebdb67 |
|
24-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Revert r8476 "Set decoder output frequency in AudioDecoder::Decode call" This change uncovered issue 4143, evading the Memcheck suppression since the signature is changed in the Decode function. A fix for this is in the making; see https://review.webrtc.org/36309004. This CL will be re-landed once the fix is in place. BUG=4143 TBR=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42089004 Cr-Commit-Position: refs/heads/master@{#8488} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8488 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
b9c18d56438eefb71ff68d47880d2b49fd380bc7 |
|
24-Feb-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Set decoder output frequency in AudioDecoder::Decode call This CL changes the way the decoder sample rate is set and updated. In practice, it only concerns the iSAC (float) codec. One single iSAC decoder instance is used for both wideband and super-wideband decoding, and the instance must be told to switch output frequency if the payload type changes. This used to be done through a call to UpdateDecoderSampleRate, but is now instead done in the Decode call as an extra parameter. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34349004 Cr-Commit-Position: refs/heads/master@{#8476} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8476 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
2c1bcf2cb4a9e19a337e52fd576242e04168d5e9 |
|
17-Feb-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
Adding decoded_fec_rate to NetEQ Network Statistics. A statistic is introduced to reflect the actual benefits of Opus FEC. It shows what percentage of samples in the rendered audio come from FEC data. BUG=3867 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34969004 Cr-Commit-Position: refs/heads/master@{#8384} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8384 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
648f5d6dc7598543e4f980cff8cca60234a7d83d |
|
10-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
pcm16b: Make input arrays const and use uint8_t[] for byte arrays There were both uint8 and uint16 versions of the pcm16b encode and decode functions; this patch removes the latter. BUG=909 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34139004 Cr-Commit-Position: refs/heads/master@{#8309} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8309 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
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e04a93bcf5e1b608c798a6a3148224b8035f0119 |
|
09-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798 and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799 Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
3800e13a3a7031220e2d21990858d4d08581e393 |
|
03-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Revert r7798 ("Move the AudioDecoder interface out of NetEq") Apparently, it caused all sorts of problems I don't have time to straighten out right now. TBR=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
00ba1a7dfd66e096ee5fb5e4e084c5565738426f |
|
03-Dec-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Move the AudioDecoder interface out of NetEq It belongs with the codecs, next to the AudioEncoder interface. R=henrik.lundin@webrtc.org, kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
6de75ca3eddf26a771143f3afdc5b0aefead505c |
|
04-Nov-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove the useless dummy state parameter to WebRtcPcm16b_DecodeW16 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7610 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
721ef633d04d86882cf935179dc37a45f539ef47 |
|
04-Nov-2014 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Remove the codec_type_ member from AudioDecoder It isn't actually required, as evidenced by the comparative ease with which it can be removed. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7606 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
3c0aae17f0e3a70fe90ecc6835926b66a3de18fb |
|
04-Sep-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Change gflags and gmock includes to be full paths. This will fix PRESUBMIT warnings developers will get due to r7014 and r7020. Also some minor style cleanup in: webrtc/modules/audio_coding/main/test/RTPFile.cc webrtc/modules/audio_coding/neteq/test/RTPjitter.cc BUG= R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
|
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/mock/mock_external_decoder_pcm16b.h
|
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/mock/mock_external_decoder_pcm16b.h
|
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/mock/mock_external_decoder_pcm16b.h
|