| 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/sinusoidal_linear_chirp_source.cc
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| f045e4da43e671ae511aa1d9b6ef2968256a745d |
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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/common_audio/resampler/sinusoidal_linear_chirp_source.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/sinusoidal_linear_chirp_source.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/sinusoidal_linear_chirp_source.cc
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| 8fc05feed43c702eb84fc26b36aecce33622b06b |
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26-Apr-2013 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add a push-based wrapper around SincResampler.
Includes a unittest to ensure we meet the same quality thresholds as SincResampler (modulo quantization error).
BUG=webrtc:1395
Review URL: https://webrtc-codereview.appspot.com/1323011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3909 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
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