fb389e3b02
Patch Set 1: Removing blanks at end of lines. Patch Set 2: Removing tabs. Patch Set 3: Fixing include-guards. Patch Set 4-7: Formatting files in the list. Patch Set 8: Formatting CNG. Patch Set 9: * Fixing comments from code review * Fixing formating in acm_dtmf_playout.cc * Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing. * Refactored constructor of ACMGenericCodec. Rest of file still to be fixed. * Fixing break; after return ...; in several files. Patch Set 10: * Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc NOTE! Not all files have the right format. That work will continue in separate CLs. Review URL: http://webrtc-codereview.appspot.com/175002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
91 lines
2.2 KiB
C++
91 lines
2.2 KiB
C++
/*
|
|
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include <string.h>
|
|
|
|
#include "acm_resampler.h"
|
|
#include "critical_section_wrapper.h"
|
|
#include "resampler.h"
|
|
#include "signal_processing_library.h"
|
|
#include "trace.h"
|
|
|
|
namespace webrtc
|
|
{
|
|
|
|
ACMResampler::ACMResampler():
|
|
|
|
_resamplerCritSect(*CriticalSectionWrapper::CreateCriticalSection())
|
|
{
|
|
}
|
|
|
|
ACMResampler::~ACMResampler()
|
|
{
|
|
|
|
delete &_resamplerCritSect;
|
|
}
|
|
|
|
|
|
WebRtc_Word16
|
|
ACMResampler::Resample10Msec(
|
|
const WebRtc_Word16* inAudio,
|
|
WebRtc_Word32 inFreqHz,
|
|
WebRtc_Word16* outAudio,
|
|
WebRtc_Word32 outFreqHz,
|
|
WebRtc_UWord8 numAudioChannels)
|
|
{
|
|
|
|
CriticalSectionScoped cs(_resamplerCritSect);
|
|
|
|
if(inFreqHz == outFreqHz)
|
|
{
|
|
memcpy(outAudio, inAudio, (inFreqHz*numAudioChannels / 100) * sizeof(WebRtc_Word16));
|
|
return (WebRtc_Word16)(inFreqHz / 100);
|
|
}
|
|
|
|
int maxLen = 480 * numAudioChannels; //max number of samples for 10ms at 48kHz
|
|
int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
|
|
int outLen;
|
|
|
|
WebRtc_Word32 ret;
|
|
ResamplerType type;
|
|
type = (numAudioChannels == 1)? kResamplerSynchronous:kResamplerSynchronousStereo;
|
|
|
|
ret = _resampler.ResetIfNeeded(inFreqHz,outFreqHz,type);
|
|
if (ret < 0)
|
|
{
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
|
|
"Error in reset of resampler");
|
|
return -1;
|
|
}
|
|
|
|
ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
|
|
if (ret < 0 )
|
|
{
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
|
|
"Error in resampler: resampler.Push");
|
|
return -1;
|
|
}
|
|
|
|
WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
|
|
|
|
return outAudioLenSmpl;
|
|
|
|
}
|
|
|
|
void
|
|
ACMResampler::SetUniqueId(
|
|
WebRtc_Word32 id)
|
|
{
|
|
CriticalSectionScoped lock(_resamplerCritSect);
|
|
_id = id;
|
|
}
|
|
|
|
} // namespace webrtc
|