git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/audio_coding/main/source/acm_resampler.cc
+++ b/modules/audio_coding/main/source/acm_resampler.cc
@@ -0,0 +1,90 @@
+/*
+ *  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