3c652b67468d182bd36aee4c31557621be50cc92 |
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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/cng/audio_encoder_cng_unittest.cc
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288886b2ec9a2dac730f115e9c3079d8439efe60 |
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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/cng/audio_encoder_cng_unittest.cc
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e9e7896293747b2084e7f018675276f5096cdafc |
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09-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Turn webrtc::Vad into a pure virtual interface Review URL: https://codereview.webrtc.org/1317243005 Cr-Commit-Position: refs/heads/master@{#9899}
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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12cfc9b4dacd6942377df1f29a64bdbec591920e |
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08-Sep-2015 |
kwiberg <kwiberg@webrtc.org> |
Fold AudioEncoderMutable into AudioEncoder It makes more sense to combine the two interfaces, since there wasn't a clear line separating them. The result is a combined interface with just over a dozen methods, half of which need to be implemented by every subclass, while the other half have sensible (and trivial) default implementations and are implemented only by the few subclasses that need non-default behavior. Review URL: https://codereview.webrtc.org/1322973004 Cr-Commit-Position: refs/heads/master@{#9894}
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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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/codecs/cng/audio_encoder_cng_unittest.cc
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b7e5054414ff524f9db81dab7917729b8c4c8bcb |
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11-Jun-2015 |
Peter Kasting <pkasting@google.com> |
Match existing type usage better. This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example: * Change a few type declarations to better match how the majority of code uses those objects. * Eliminate "< 0" check for unsigned values. * Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar. * Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects. * Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t. * Similarly, add casts when passing a larger type to a function taking a smaller one. * Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar. * Use "false" instead of "0" for setting a bool. * Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t. BUG=none R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org TBR=andrew, asapersson, henrika Review URL: https://codereview.webrtc.org/1168753002 Cr-Commit-Position: refs/heads/master@{#9419}
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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367c868c998e96bc1aac41b607548d6125fa6b1e |
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22-May-2015 |
Henrik Lundin <henrik.lundin@webrtc.org> |
AudioEncoderCng: Handle case where speech encoder is reset Previously, AudioEncoderCng required the speech encoder to not change its mind regarding the number of 10 ms frames in the next packet between calls to AudioEncoderCng::EncodeInternal()---specifically, it could handle an upward but not a downward adjustment. With this patch, it can handle a downward adjustment too, by simply saving the overshoot data for the next call to EncodeInternal(). It will still not handle the case where the encoder's reported number of 10 ms frames in the next packet is inconsistent with the behavior of its Encode() function when called with no intervening changes to the encoder. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/53469005 Cr-Commit-Position: refs/heads/master@{#9261}
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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9afaee74ab1ef36c8b4ea4c22f4c5aebf2359da2 |
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19-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() Old review at: https://webrtc-codereview.appspot.com/43839004/ R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45769004 Cr-Commit-Position: refs/heads/master@{#8788} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8788 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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019955d77015fed0b2dcec0cc62a8bdd63e0481e |
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18-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8749 "We changed Encode() and EncodeInternal() return typ..." The reason is that this cl adds a static initializer so we can't roll webrtc into Chromium. See audio_encoder.cc and 'sizes' regression here: http://build.chromium.org/p/chromium/builders/Linux%20x64/builds/186 > We changed Encode() and EncodeInternal() return type from bool to void in this issue: > https://webrtc-codereview.appspot.com/38279004/ > Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info. > > R=kwiberg@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/43839004 TBR=jmarusic@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49449004 Cr-Commit-Position: refs/heads/master@{#8772} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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0cb612b43bc1ef42cde8cb3887dc48917d5a58dd |
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17-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
We changed Encode() and EncodeInternal() return type from bool to void in this issue: https://webrtc-codereview.appspot.com/38279004/ Now we don't have to pass EncodedInfo as output parameter, but can return it instead. This also adds the benefit of making clear that EncodeInternal() needs to fill in this info. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43839004 Cr-Commit-Position: refs/heads/master@{#8749} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8749 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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51ccf376387266225cd8c78e63238b725860f0af |
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10-Mar-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: add method MaxEncodedBytes Added method AudioEncoder::MaxEncodedBytes() and provided implementations in derived encoders. This method returns the number of bytes that can be produced by the encoder at each Encode() call. Unit tests were updated to use the new method. Buffer allocation was not changed in AudioCodingModuleImpl::Encode(). It will be done after additional investigation. Other refactoring work that was done, that may not be obvious why: 1. Moved some code into AudioEncoderCng::EncodePassive() to make it more consistent with EncodeActive(). 2. Changed the order of NumChannels() and RtpTimestampRateHz() declarations in AudioEncoderG722 and AudioEncoderCopyRed classes. It just bothered me that the order was not the same as in AudioEncoder class and its other derived classes. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40259005 Cr-Commit-Position: refs/heads/master@{#8671} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8671 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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c86bbbaa9348b868e94c021426abcc2f5e0144b0 |
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04-Mar-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add speech flag to EncodedInfo The flag indicates if the encoded bitstream is speech or comfort noise. COAUTHOR=kwiberg@webrtc.org R=jmarusic@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42629004 Cr-Commit-Position: refs/heads/master@{#8598} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8598 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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14665ff7d4024d07e58622f498b23fd980001871 |
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04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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b1f0de30be3397eba3d423b71abc5c50db2a1665 |
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26-Feb-2015 |
jmarusic@webrtc.org <jmarusic@webrtc.org> |
AudioEncoder: change Encode and EncodeInternal return type to void After code cleanup done on issues: https://webrtc-codereview.appspot.com/34259004/ https://webrtc-codereview.appspot.com/43409004/ https://webrtc-codereview.appspot.com/34309004/ https://webrtc-codereview.appspot.com/34309004/ https://webrtc-codereview.appspot.com/36209004/ https://webrtc-codereview.appspot.com/40899004/ https://webrtc-codereview.appspot.com/39279004/ https://webrtc-codereview.appspot.com/42099005/ and the similar work done for AudioEncoderDecoderIsacT, methods AudioEncoder::Encode and AudioEncoder::EncodeInternal will always succeed. Therefore, there is no need for them to return bool value that represents success or failure. R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38279004 Cr-Commit-Position: refs/heads/master@{#8518} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8518 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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00b8f6b3643332cce1ee711715f7fbb824d793ca |
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26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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05211277798ca4791fbdc508e24d7fd06d5ee6ff |
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18-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
AudioEncoder: Rename virtual accessors to CamelCase Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz() are simple accessors for almost all implementations of AudioEncoder, they are virtual and not guaranteed to be just simple accessors. Thus, it makes more sense to use the normal CamelCase naming scheme. BUG=4235 R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34239004 Cr-Commit-Position: refs/heads/master@{#8407} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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8bb32d600b85b2d3b85598f43f009f82faad6006 |
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27-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Minor updates to AudioEncoderCng Removing sample_rate_hz_ from AudioEncoderCng and from the config struct. The sample rate will now be read from the underlying speech codec. BUG=3926 COAUTHOR:kwiberg@webrtc.org R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40559004 Cr-Commit-Position: refs/heads/master@{#8173} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8173 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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478cedc055f95bd160b53a4d7b69d8b3dd023ec7 |
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27-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Add new methods to AudioEncoder interface The following three methods are added: rtp_timestamp_rate_hz() SetTargetBitrate() SetProjectedPacketLossRate() Default implementations are provided, and a few overrides are implemented. AudioEncoderCopyRed and AudioEncoderCng propagate the new methods to the underlying speech codec. BUG=3926 COAUTHOR:kwiberg@webrtc.org R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34049004 Cr-Commit-Position: refs/heads/master@{#8171} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8171 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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3b79daff14127f3adb19b16d94336d44ff49e841 |
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12-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Moving encoded_bytes into EncodedInfo BUG=3926 R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
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ff1a3e36bdd023fc2d3bef9af6b161ce144ffd6b |
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10-Dec-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Make an AudioEncoder subclass for comfort noise BUG=3926 R=bjornv@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
|