14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
73 lines
2.3 KiB
C++
73 lines
2.3 KiB
C++
/*
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* Copyright (c) 2012 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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#include <string.h>
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#include "acm_resampler.h"
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#include "critical_section_wrapper.h"
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#include "resampler.h"
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#include "signal_processing_library.h"
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#include "trace.h"
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namespace webrtc {
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ACMResampler::ACMResampler()
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: _resamplerCritSect(CriticalSectionWrapper::CreateCriticalSection()) {
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}
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ACMResampler::~ACMResampler() {
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delete _resamplerCritSect;
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}
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WebRtc_Word16 ACMResampler::Resample10Msec(const WebRtc_Word16* inAudio,
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WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels) {
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CriticalSectionScoped cs(_resamplerCritSect);
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if (inFreqHz == outFreqHz) {
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size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
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memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
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return static_cast<WebRtc_Word16>(inFreqHz / 100);
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}
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// |maxLen| is maximum number of samples for 10ms at 48kHz.
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int maxLen = 480 * numAudioChannels;
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int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
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int outLen;
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WebRtc_Word32 ret;
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ResamplerType type;
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type = (numAudioChannels == 1) ? kResamplerSynchronous :
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kResamplerSynchronousStereo;
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ret = _resampler.ResetIfNeeded(inFreqHz, outFreqHz, type);
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if (ret < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in reset of resampler");
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return -1;
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}
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ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
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if (ret < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in resampler: resampler.Push");
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return -1;
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}
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WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
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return outAudioLenSmpl;
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}
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} // namespace webrtc
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