As part of the review, refactored AudioConverter into internal derived
classes, each focused on one type of conversion. A factory method
returns the correct converter (or chain of converters, via
CompositionConverter).
BUG=b/18938079
R=rojer@google.com
Review URL: https://webrtc-codereview.appspot.com/35699004
Cr-Commit-Position: refs/heads/master@{#8322}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8322 4adac7df-926f-26a2-2b94-8c16560cd09d
This changes some method signatures to better reflect how callers are actually
using them. This also has the tendency to make signatures more consistent about
e.g. using int (instead of int16_t) for lengths of things like vectors, and
using int16_t (instead of int) for e.g. counts of bits in a value.
This also removes a couple of functions that were only called in unittests.
BUG=3353,chromium:81439
TEST=none
R=andrew@webrtc.org, bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
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
This may fix a Dr. Memory error: "allocated with operator new, freed
with operator delete[]". I suspect this is a false positive; in the
existing implementation the reset causes a delete[] on NULL. This is
a no-op of course, but Dr. Memory might be flagging it. We shall see.
In any case, this change is an improvement.
BUG=2321
TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/2215004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4748 4adac7df-926f-26a2-2b94-8c16560cd09d
Ideally, PushSincResampler would have very little overhead on
SincResampler. This gets closer to that ideal.
Replace std::min/max and floor with inline functions. Add a benchmark
test to verify the improvement.
On a MacBook Retina, this results in PushSincResampler::Resample()
accounting for ~1% of CPU usage on voe_cmd_test vs the earlier ~2%
(with ISAC16 and 48 kHz audio devices).
Using the new benchmark, this results in a performance improvement of:
16 -> 44.1 : 1.7x
16 -> 48 : 1.9x
32 -> 44.1 : 1.6x
32 -> 48 : 1.7x
44.1 -> 16 : 1.5x
44.1 -> 32 : 1.7x
44.1 -> 48 : 1.7x
48 -> 16 : 1.5x
48 -> 32 : 1.5x
48 -> 44.1 : 1.8x
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2157005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4695 4adac7df-926f-26a2-2b94-8c16560cd09d
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
* 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
* Brings in on-the-fly sample ratio updates (or varispeed) with minor modifications to build in webrtc.
* Moved SSE and NEON optimized functions into their own files to handle run-time detection properly. NEON optimizations now enabled.
TESTED=unit tests and ran voe_cmd_test loopback with both devices using 44.1 kHz to exercise SincResampler in real-time.
R=dalecurtis@chromium.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1438004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3987 4adac7df-926f-26a2-2b94-8c16560cd09d
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
This is the first in a series of CLs to bring arbitrary resampling to webrtc.
* Replace Chromium-specific helpers with their respective webrtc versions.
* Add a second constructor to permit runtime selection of block_size.
* Add stringize_macros to system_wrappers.
BUG=webrtc:1395
TESTED=unit tests
Review URL: https://webrtc-codereview.appspot.com/1097012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
