6955870806624479723addfae6dcf5d13968796c |
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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/common_audio/resampler/push_resampler.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/common_audio/resampler/push_resampler.cc
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f5a33f145b74d9c6058c670baf7b6201b78f6e48 |
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19-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Resampler modifications in preparation for arbitrary audioproc rates. - Templatize PushResampler to support int16 and float. - Add a helper method to PushSincResampler to compute the algorithmic delay. This is a prerequisite of: http://review.webrtc.org/9919004/ BUG=2894 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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31628aae7e0d5a00e816f1f5db4b65101319a307 |
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22-Oct-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Upgrade scoped_ptr to Chromium's latest version. Analogous to the recent libjingle change: http://cl/54929753-p10. This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather than scoped_array and scoped_ptr_malloc respectively. - Add Chromium's template-based COMPILE_ASSERT. We didn't have this previously in order to support the macro in C. Instead, move the existing macro to compile_assert_c.h. - Additionally copy the move.h and template_util.h depedencies and add the WARN_UNUSED_RESULT macro. - Leave scoped_array and scoped_ptr_malloc for now, but mark as deprecated. - Remove scoped_ptr foo(NULL) use. The default constructor handles it. - Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc. - Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove some repeated code. TESTED=trybots R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2449005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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12dc1a38ca54a000e4fecfbc6d41138b895c9ca5 |
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05-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch C++-style C headers with their C equivalents. The C++ headers define the C functions within the std:: namespace, but we mainly don't use the std:: namespace for C functions. Therefore we should include the C headers. BUG=1833 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1917004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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b86fbaf1d41db539205ec671ff399a3a3aa50734 |
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26-Jul-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Downstream latest Chromium SincResampler changes. Replace the BlockSize() workaround we were using previously to support the push wrapper with the upstream request_frames interface. This requires a bit of a trick to ensure we don't add more delay than necessary. On the first pass we use a dummy Resample() call in order to prime the buffer such that all later calls only require a single input request through Run(). Notably, this brings in an optimized loop condition, improving performance by ~2% - 3% on tested platforms and avoids a 20% performance hit with clang. This addresses issue2041. Only negligible changes to the PushSincResamplerTest SNR thresholds, due to a fractional sample adjustment in output delay. This still retains the per-instance CPU detection, as webrtc lacks a LazyInstance helper for static initialization. Ideally, we would adopt SetRatio() in PushSincResampler's InitializeIfNeeded() for on-the-fly changes, but this will require a way to update request_frames. The diff against Chromium upstream is available here: https://codereview.chromium.org/19470003 BUG=2041 TESTED=unit tests, voe_cmd_test in loopback running through all codecs with 44.1 kHz and 48 kHz device formats using a stereo mic. R=dalecurtis@chromium.org Review URL: https://webrtc-codereview.appspot.com/1838004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4406 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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c1eb560a5cbfea2a55e6845a967776dbaae7ba01 |
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03-Jun-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace the old resampler with SincResampler in the voice engine signal path. * The old resampler was found to have a wraparound bug. * Remove support for the old resampler from PushResampler. * Use PushResampler in AudioCodingModule. * The old resampler must still be removed from the file utility. BUG=webrtc:1867,webrtc:827 TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio R=henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1590004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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50b2efef6ecb51a9d5818345c58533c5d236ec29 |
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29-Apr-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a wrapper around PushSincResampler and the old Resampler. The old resampler is used whenever it supports the requested rates. Otherwise the sinc resampler is enabled. Integrated with output_mixer in order to test the change through output_mixer_unittest. The sinc resampler will not yet be used, since we don't feed VoE with any rates that trigger it. BUG=webrtc:1395 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1355004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/push_resampler.cc
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