b159c2e3dd
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
48 lines
1.8 KiB
C++
48 lines
1.8 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#define WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Clamp the floating |value| to the range representable by an int16_t.
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static inline float ClampInt16(float value) {
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const float kMaxInt16 = 32767.f;
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const float kMinInt16 = -32768.f;
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return value < kMinInt16 ? kMinInt16 :
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(value > kMaxInt16 ? kMaxInt16 : value);
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}
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// Return a rounded int16_t of the floating |value|. Doesn't handle overflow;
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// use ClampInt16 if necessary.
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static inline int16_t RoundToInt16(float value) {
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return static_cast<int16_t>(value < 0.f ? value - 0.5f : value + 0.5f);
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}
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// Deinterleave audio from |interleaved| to the channel buffers pointed to
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// by |deinterleaved|. There must be sufficient space allocated in the
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// |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
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// per buffer).
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void Deinterleave(const int16_t* interleaved, int samples_per_channel,
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int num_channels, int16_t** deinterleaved);
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// Interleave audio from the channel buffers pointed to by |deinterleaved| to
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// |interleaved|. There must be sufficient space allocated in |interleaved|
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// (|samples_per_channel| * |num_channels|).
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void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
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int num_channels, int16_t* interleaved);
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
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