Speex 回声消除

为什么需要声学回声消除呢？在一般的VOIP软件或视频会议系统中，假设我们只有A和B两个人在通话，首先，A的声音传给B，B然后用喇叭放出来，而这时B的MIC呢则会采集到喇叭放出来的声音，然后传回给A，如果这个传输的过程中时延足够大，A就会听到一个和自己刚才说过的话一样的声音，这就是回声，声学回声消除器的作用就是在B端对B采集到的声音进行处理，把采集到声音包含的A的声音去掉再传给A，这样，A就不会听到自己说过的话了。
声学回声消除的原理我就不说了，这在网上有很多文档，网上缺少的是实现，所以，我在这把一个开源的声学回声消除器介绍一下，希望对有些有人用，如果有人知道怎么把这消除器用的基于实时流的VOIP软件中，希望能一起分享一下。
这个声学回声消除器是一个著名的音频编解码器speex中的一部分，1.1.9版本后的回声消除器才起作用，以前版本的都不行，我用的也是这个版本，测试表明，用同一个模拟文件，它有效果比INTEL IPP库4.1版中的声学回声消除器的还要好。
先说编译。首先，从www.speex.org上下载speex1.1.9的源代码，解压，打开speex\win32\libspeex中的libspeex.dsw，这个工作区里有两个工程，一个是 libspeex，另一个是libspeex_dynamic。然后，将libspeex中的mdf.c文件添加到工程libspeex中，编译即可。
以下是我根据文档封装的一个类，里面有一个测试程序： //file name: speexEC.h
#ifndef SPEEX_EC_H
#define SPEEX_EC_H
#include <stdio.h>
#include <stdlib.h>
#include "speex/speex_echo.h"
#include "speex/speex_preprocess.h"
class CSpeexEC
{
public:
CSpeexEC();
~CSpeexEC();
void Init(int frame_size=160, int filter_length=1280, int sampling_rate=8000);
void DoAEC(short *mic, short *ref, short *out);

protected:
void Reset();

private:
bool          m_bHasInit;
SpeexEchoState*    m_pState;
      SpeexPreprocessState* m_pPreprocessorState;
int          m_nFrameSize;
int          m_nFilterLen;
int          m_nSampleRate;
float*          m_pfNoise;
};

#endif

//fine name:speexEC.cpp
#include "SpeexEC.h"

CSpeexEC::CSpeexEC()
{
m_bHasInit    = false;
m_pState    = NULL;
m_pPreprocessorState  = NULL;
m_nFrameSize    = 160;
m_nFilterLen    = 160*8;
m_nSampleRate    = 8000;
m_pfNoise    = NULL;
}

CSpeexEC::~CSpeexEC()
{
Reset();
}

void CSpeexEC::Init(int frame_size, int filter_length, int sampling_rate)
{
Reset();

if (frame_size<=0 || filter_length<=0 || sampling_rate<=0)
{
  m_nFrameSize  =160;
  m_nFilterLen  = 160*8;
  m_nSampleRate = 8000;
}
else
{
  m_nFrameSize  =frame_size;
  m_nFilterLen  = filter_length;
  m_nSampleRate = sampling_rate;
}

m_pState = speex_echo_state_init(m_nFrameSize, m_nFilterLen);
m_pPreprocessorState = speex_preprocess_state_init(m_nFrameSize, m_nSampleRate);
m_pfNoise = new float[m_nFrameSize+1];
m_bHasInit = true;
}

void CSpeexEC::Reset()
{
if (m_pState != NULL)
{
  speex_echo_state_destroy(m_pState);
  m_pState = NULL;
}
if (m_pPreprocessorState != NULL)
{
  speex_preprocess_state_destroy(m_pPreprocessorState);
  m_pPreprocessorState = NULL;
}
if (m_pfNoise != NULL)
{
  delete []m_pfNoise;
  m_pfNoise = NULL;
}
m_bHasInit = false;
}

void CSpeexEC:DoAEC(short* mic, short* ref, short* out)
{
if (!m_bHasInit)
  return;

speex_echo_cancel(m_pState, mic, ref, out, m_pfNoise);
      speex_preprocess(m_pPreprocessorState, (__int16 *)out, m_pfNoise);
   
}

可以看出，这个回声消除器类很简单，只要初始化一下就可以调用了。但是，要注意的是，传给回声消除器的两个声音信号，必须同步得非常的好，就是说，在B端，接收到A说的话以后，要把这些话音数据传给回声消除器做参考，然后再传给声卡，声卡再放出来，这有一段延时，这时，B再采集，然后传给回声消除器，与那个参考数据比较，从采集到的数据中把频域和参考数据相同的部分消除掉。如果传给消除器的两个信号同步得不好，即两个信号找不到频域相同的部分，就没有办法进行消除了。
测试程序：

#define NN 160
void main()
{
FILE* ref_fd, *mic_fd, *out_fd;
short ref[NN], mic[NN], out[NN];
ref_fd = fopen ("ref.pcm", "rb");　//打开参考文件，即要消除的声音
mic_fd = fopen ("mic.pcm",  "rb");//打开mic采集到的声音文件，包含回声在里面
out_fd = fopen ("echo.pcm", "wb");//消除了回声以后的文件

CSpeexEC ec;
ec.Init();

while (fread(mic, 1, NN*2, mic_fd))
    {
          fread(ref, 1, NN*2, ref_fd);  
          ec.DoAEC(mic, ref, out);
          fwrite(out, 1, NN*2, out_fd);
    }
 
    fclose(ref_fd);
    fclose(mic_fd);
    fclose(out_fd);
}

　　以上的程序是用文件来模拟回声和MIC，但在实时流中是大不一样的，在一般的VOIP软件中，接收对方的声音并传到声卡中播放是在一个线程中进行的，而采集本地的声音并传送到对方又是在另一个线程中进行的，而声学回声消除器在对采集到的声音进行回声消除的同时，还需要播放线程中的数据作为参考，而要同步这两个线程中的数据是非常困难的，因为稍稍有些不同步，声学回声消除器中的自适应滤波器就会发散，不但消除不了回声，还会破坏原始采集到的声音，使被破坏的声音难以分辨。我做过好多尝试，始终无法用软件来实现对这两个线程中的数据进行同步，导致实现失败，希望有经验的网友们一起分享一下这方面的经验。



示例代码：


Sample code

This section shows sample code for encoding and decoding speech using the Speex API. The commands can be used to encode and decode a file by calling:
% sampleenc in_file.sw | sampledec out_file.sw
where both files are raw (no header) files encoded at 16 bits per sample (in the machine natural endianness).

sampleenc.c

sampleenc takes a raw 16 bits/sample file, encodes it and outputs a Speex stream to stdout. Note that the packing used is NOT compatible with that of speexenc/speexdec.


#include <speex/speex.h>
#include <stdio.h>

#define FRAME_SIZE 160
int main(int argc, char **argv)
{
char *inFile;
FILE *fin;
short in[FRAME_SIZE];
float input[FRAME_SIZE];
char cbits[200];
int nbBytes;

void *state;

SpeexBits bits;
int i, tmp;

state = speex_encoder_init(&speex_nb_mode);

tmp=8;
speex_encoder_ctl(state, SPEEX_SET_QUALITY, &tmp);
inFile = argv[1];
fin = fopen(inFile, "r");

speex_bits_init(&bits);
while (1)
{

fread(in, sizeof(short), FRAME_SIZE, fin);
if (feof(fin))
break;

for (i=0;i<FRAME_SIZE;i++)
input[i]=in[i];

speex_bits_reset(&bits);

speex_encode(state, input, &bits);

nbBytes = speex_bits_write(&bits, cbits, 200);

fwrite(&nbBytes, sizeof(int), 1, stdout);

fwrite(cbits, 1, nbBytes, stdout);
}

speex_encoder_destroy(state);

speex_bits_destroy(&bits);
fclose(fin);
return 0;
}

sampledec.c

sampledec reads a Speex stream from stdin, decodes it and outputs it to a raw 16 bits/sample file. Note that the packing used is NOT compatible with that of speexenc/speexdec.


#include <speex/speex.h>
#include <stdio.h>

#define FRAME_SIZE 160
int main(int argc, char **argv)
{
char *outFile;
FILE *fout;

short out[FRAME_SIZE];

float output[FRAME_SIZE];
char cbits[200];
int nbBytes;

void *state;

SpeexBits bits;
int i, tmp;

state = speex_decoder_init(&speex_nb_mode);

tmp=1;
speex_decoder_ctl(state, SPEEX_SET_ENH, &tmp);
outFile = argv[1];
fout = fopen(outFile, "w");

speex_bits_init(&bits);
while (1)
{

fread(&nbBytes, sizeof(int), 1, stdin);
fprintf (stderr, "nbBytes: %d\n", nbBytes);
if (feof(stdin))
break;

fread(cbits, 1, nbBytes, stdin);

speex_bits_read_from(&bits, cbits, nbBytes);

speex_decode(state, &bits, output);

for (i=0;i<FRAME_SIZE;i++)
out[i]=output[i];

fwrite(out, sizeof(short), FRAME_SIZE, fout);
}

speex_decoder_destroy(state);

speex_bits_destroy(&bits);
fclose(fout);
return 0;
}




 


 




开源 H323 协议中封装的使用参考代码：




#include <ptlib.h>

#ifdef __GNUC__
#pragma implementation "speexcodec.h"
#endif

#include "speexcodec.h"

#include "h323caps.h"
#include "h245.h"
#include "rtp.h"

extern "C" {
#include "speex/libspeex/speex.h"
};


#define new PNEW

#define XIPH_COUNTRY_CODE            0xB5  // (181) Country code for United States
#define XIPH_T35EXTENSION            0
#define XIPH_MANUFACTURER_CODE  0x0026 // Allocated by Delta Inc

#define EQUIVALENCE_COUNTRY_CODE            9  // Country code for Australia
#define EQUIVALENCE_T35EXTENSION            0
#define EQUIVALENCE_MANUFACTURER_CODE  61 // Allocated by Australian Communications Authority, Oct 2000

#define SAMPLES_PER_FRAME              160

#define SPEEX_BASE_NAME "Speex"

#define SPEEX_NARROW2_H323_NAME      SPEEX_BASE_NAME "Narrow-5.95k{sw}"
#define SPEEX_NARROW3_H323_NAME      SPEEX_BASE_NAME "Narrow-8k{sw}"
#define SPEEX_NARROW4_H323_NAME      SPEEX_BASE_NAME "Narrow-11k{sw}"
#define SPEEX_NARROW5_H323_NAME      SPEEX_BASE_NAME "Narrow-15k{sw}"
#define SPEEX_NARROW6_H323_NAME      SPEEX_BASE_NAME "Narrow-18.2k{sw}"

H323_REGISTER_CAPABILITY(SpeexNarrow2AudioCapabil ity, SPEEX_NARROW2_H323_NAME);
H323_REGISTER_CAPABILITY(SpeexNarrow3AudioCapabil ity, SPEEX_NARROW3_H323_NAME);
H323_REGISTER_CAPABILITY(SpeexNarrow4AudioCapabil ity, SPEEX_NARROW4_H323_NAME);
H323_REGISTER_CAPABILITY(SpeexNarrow5AudioCapabil ity, SPEEX_NARROW5_H323_NAME);
H323_REGISTER_CAPABILITY(SpeexNarrow6AudioCapabil ity, SPEEX_NARROW6_H323_NAME);

#define XIPH_SPEEX_NARROW2_H323_NAME      SPEEX_BASE_NAME "Narrow-5.95k(Xiph){sw}"
#define XIPH_SPEEX_NARROW3_H323_NAME      SPEEX_BASE_NAME "Narrow-8k(Xiph){sw}"
#define XIPH_SPEEX_NARROW4_H323_NAME      SPEEX_BASE_NAME "Narrow-11k(Xiph){sw}"
#define XIPH_SPEEX_NARROW5_H323_NAME      SPEEX_BASE_NAME "Narrow-15k(Xiph){sw}"
#define XIPH_SPEEX_NARROW6_H323_NAME      SPEEX_BASE_NAME "Narrow-18.2k(Xiph){sw}"

H323_REGISTER_CAPABILITY(XiphSpeexNarrow2AudioCap ability, XIPH_SPEEX_NARROW2_H323_NAME);
H323_REGISTER_CAPABILITY(XiphSpeexNarrow3AudioCap ability, XIPH_SPEEX_NARROW3_H323_NAME);
H323_REGISTER_CAPABILITY(XiphSpeexNarrow4AudioCap ability, XIPH_SPEEX_NARROW4_H323_NAME);
H323_REGISTER_CAPABILITY(XiphSpeexNarrow5AudioCap ability, XIPH_SPEEX_NARROW5_H323_NAME);
H323_REGISTER_CAPABILITY(XiphSpeexNarrow6AudioCap ability, XIPH_SPEEX_NARROW6_H323_NAME);

/

static int Speex_Bits_Per_Second(int mode) {
      void *tmp_coder_state;
      int bitrate;
      tmp_coder_state = speex_encoder_init(&speex_nb_mode);
      speex_encoder_ctl(tmp_coder_state, SPEEX_SET_QUALITY, &mode);
      speex_encoder_ctl(tmp_coder_state, SPEEX_GET_BITRATE, &bitrate);
      speex_encoder_destroy(tmp_coder_state);
      return bitrate;
}

static int Speex_Bytes_Per_Frame(int mode) {
      int bits_per_frame = Speex_Bits_Per_Second(mode) / 50; // (20ms frame size)
      return ((bits_per_frame+7)/8); // round up
}

OpalMediaFormat const OpalSpeexNarrow_5k95(OPAL_SPEEX_NARROW_5k95,
                                                                                    OpalMediaFormat::DefaultAudioSessionID,
                                                                                    RTP_DataFrame::DynamicBase,
                                                                                    TRUE,  // Needs jitter
                                                                                    Speex_Bits_Per_Second(2),
                                                                                    Speex_Bytes_Per_Frame(2),
                                                                                    SAMPLES_PER_FRAME, // 20 milliseconds
                                                                                    OpalMediaFormat::AudioTimeUnits);

OpalMediaFormat const OpalSpeexNarrow_8k(OPAL_SPEEX_NARROW_8k,
                                                                                OpalMediaFormat::DefaultAudioSessionID,
                                                                                RTP_DataFrame::DynamicBase,
                                                                                TRUE,  // Needs jitter
                                                                                Speex_Bits_Per_Second(3),
                                                                                Speex_Bytes_Per_Frame(3),
                                                                                SAMPLES_PER_FRAME, // 20 milliseconds
                                                                                OpalMediaFormat::AudioTimeUnits);

OpalMediaFormat const OpalSpeexNarrow_11k(OPAL_SPEEX_NARROW_11k,
                                                                                  OpalMediaFormat::DefaultAudioSessionID,
                                                                                  RTP_DataFrame::DynamicBase,
                                                                                  TRUE,  // Needs jitter
                                                                                  Speex_Bits_Per_Second(4),
                                                                                  Speex_Bytes_Per_Frame(4),
                                                                                  SAMPLES_PER_FRAME, // 20 milliseconds
                                                                                  OpalMediaFormat::AudioTimeUnits);

OpalMediaFormat const OpalSpeexNarrow_15k(OPAL_SPEEX_NARROW_15k,
                                                                                  OpalMediaFormat::DefaultAudioSessionID,
                                                                                  RTP_DataFrame::DynamicBase,
                                                                                  TRUE,  // Needs jitter
                                                                                  Speex_Bits_Per_Second(5),
                                                                                  Speex_Bytes_Per_Frame(5),
                                                                                  SAMPLES_PER_FRAME, // 20 milliseconds
                                                                                  OpalMediaFormat::AudioTimeUnits);

OpalMediaFormat const OpalSpeexNarrow_18k2(OPAL_SPEEX_NARROW_18k2,
                                                                                    OpalMediaFormat::DefaultAudioSessionID,
                                                                                    RTP_DataFrame::DynamicBase,
                                                                                    TRUE,  // Needs jitter
                                                                                    Speex_Bits_Per_Second(6),
                                                                                    Speex_Bytes_Per_Frame(6),
                                                                                    SAMPLES_PER_FRAME, // 20 milliseconds
                                                                                    OpalMediaFormat::AudioTimeUnits);


/

SpeexNonStandardAudioCap ability::SpeexNonStandardAudioCap ability(int mode)
  : H323NonStandardAudioCapa bility(1, 1,
                                                                    EQUIVALENCE_COUNTRY_CODE,
                                                                    EQUIVALENCE_T35EXTENSION,
                                                                    EQUIVALENCE_MANUFACTURER_CODE,
                                                                    NULL, 0, 0, P_MAX_INDEX)
{
  PStringStream s;
  s << "Speex bs" << speex_nb_mode.bitstream_version << " Narrow" << mode;
  PINDEX len = s.GetLength();
  memcpy(nonStandardData.GetPointer(len), (const char *)s, len);
}


/

SpeexNarrow2AudioCapabil ity::SpeexNarrow2AudioCapabil ity()
  : SpeexNonStandardAudioCap ability(2)
{
}


PObject * SpeexNarrow2AudioCapabil ity::Clone() const
{
  return new SpeexNarrow2AudioCapabil ity(*this);
}


PString SpeexNarrow2AudioCapabil ity::GetFormatName() const
{
  return SPEEX_NARROW2_H323_NAME;
}


H323Codec * SpeexNarrow2AudioCapabil ity::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_5k95, 2, direction);
}


/

SpeexNarrow3AudioCapabil ity::SpeexNarrow3AudioCapabil ity()
  : SpeexNonStandardAudioCap ability(3)
{
}


PObject * SpeexNarrow3AudioCapabil ity::Clone() const
{
  return new SpeexNarrow3AudioCapabil ity(*this);
}


PString SpeexNarrow3AudioCapabil ity::GetFormatName() const
{
  return SPEEX_NARROW3_H323_NAME;
}


H323Codec * SpeexNarrow3AudioCapabil ity::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_8k, 3, direction);
}


/

SpeexNarrow4AudioCapabil ity::SpeexNarrow4AudioCapabil ity()
  : SpeexNonStandardAudioCap ability(4)
{
}


PObject * SpeexNarrow4AudioCapabil ity::Clone() const
{
  return new SpeexNarrow4AudioCapabil ity(*this);
}


PString SpeexNarrow4AudioCapabil ity::GetFormatName() const
{
  return SPEEX_NARROW4_H323_NAME;
}


H323Codec * SpeexNarrow4AudioCapabil ity::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_11k, 4, direction);
}


/

SpeexNarrow5AudioCapabil ity::SpeexNarrow5AudioCapabil ity()
  : SpeexNonStandardAudioCap ability(5)
{
}


PObject * SpeexNarrow5AudioCapabil ity::Clone() const
{
  return new SpeexNarrow5AudioCapabil ity(*this);
}


PString SpeexNarrow5AudioCapabil ity::GetFormatName() const
{
  return SPEEX_NARROW5_H323_NAME;
}


H323Codec * SpeexNarrow5AudioCapabil ity::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_15k, 5, direction);
}


/

SpeexNarrow6AudioCapabil ity::SpeexNarrow6AudioCapabil ity()
  : SpeexNonStandardAudioCap ability(6)
{
}


PObject * SpeexNarrow6AudioCapabil ity::Clone() const
{
  return new SpeexNarrow6AudioCapabil ity(*this);
}


PString SpeexNarrow6AudioCapabil ity::GetFormatName() const
{
  return SPEEX_NARROW6_H323_NAME;
}


H323Codec * SpeexNarrow6AudioCapabil ity::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_18k2, 6, direction);
}


/

XiphSpeexNonStandardAudi oCapability::XiphSpeexNonStandardAudi oCapability(int mode)
  : H323NonStandardAudioCapa bility(1, 1,
                                                                    XIPH_COUNTRY_CODE,
                                                                    XIPH_T35EXTENSION,
                                                                    XIPH_MANUFACTURER_CODE,
                                                                    NULL, 0, 0, P_MAX_INDEX)
{
  // FIXME: To be replaced by an ASN defined block of data
  PStringStream s;
  s << "Speex bs" << speex_nb_mode.bitstream_version << " Narrow" << mode;
  PINDEX len = s.GetLength();
  memcpy(nonStandardData.GetPointer(len), (const char *)s, len);
}


/

XiphSpeexNarrow2AudioCap ability::XiphSpeexNarrow2AudioCap ability()
  : XiphSpeexNonStandardAudi oCapability(2)
{
}


PObject * XiphSpeexNarrow2AudioCap ability::Clone() const
{
  return new XiphSpeexNarrow2AudioCap ability(*this);
}


PString XiphSpeexNarrow2AudioCap ability::GetFormatName() const
{
  return XIPH_SPEEX_NARROW2_H323_NAME;
}


H323Codec * XiphSpeexNarrow2AudioCap ability::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_5k95, 2, direction);
}


/

XiphSpeexNarrow3AudioCap ability::XiphSpeexNarrow3AudioCap ability()
  : XiphSpeexNonStandardAudi oCapability(3)
{
}


PObject * XiphSpeexNarrow3AudioCap ability::Clone() const
{
  return new XiphSpeexNarrow3AudioCap ability(*this);
}


PString XiphSpeexNarrow3AudioCap ability::GetFormatName() const
{
  return XIPH_SPEEX_NARROW3_H323_NAME;
}


H323Codec * XiphSpeexNarrow3AudioCap ability::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_8k, 3, direction);
}


/

XiphSpeexNarrow4AudioCap ability::XiphSpeexNarrow4AudioCap ability()
  : XiphSpeexNonStandardAudi oCapability(4)
{
}


PObject * XiphSpeexNarrow4AudioCap ability::Clone() const
{
  return new XiphSpeexNarrow4AudioCap ability(*this);
}


PString XiphSpeexNarrow4AudioCap ability::GetFormatName() const
{
  return XIPH_SPEEX_NARROW4_H323_NAME;
}


H323Codec * XiphSpeexNarrow4AudioCap ability::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_11k, 4, direction);
}


/

XiphSpeexNarrow5AudioCap ability::XiphSpeexNarrow5AudioCap ability()
  : XiphSpeexNonStandardAudi oCapability(5)
{
}


PObject * XiphSpeexNarrow5AudioCap ability::Clone() const
{
  return new XiphSpeexNarrow5AudioCap ability(*this);
}


PString XiphSpeexNarrow5AudioCap ability::GetFormatName() const
{
  return XIPH_SPEEX_NARROW5_H323_NAME;
}


H323Codec * XiphSpeexNarrow5AudioCap ability::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_15k, 5, direction);
}


/

XiphSpeexNarrow6AudioCap ability::XiphSpeexNarrow6AudioCap ability()
  : XiphSpeexNonStandardAudi oCapability(6)
{
}


PObject * XiphSpeexNarrow6AudioCap ability::Clone() const
{
  return new XiphSpeexNarrow6AudioCap ability(*this);
}


PString XiphSpeexNarrow6AudioCap ability::GetFormatName() const
{
  return XIPH_SPEEX_NARROW6_H323_NAME;
}


H323Codec * XiphSpeexNarrow6AudioCap ability::CreateCodec(H323Codec::Direction direction) const
{
  return new SpeexCodec(OpalSpeexNarrow_18k2, 6, direction);
}


/

const float MaxSampleValue    = 32767.0;
const float MinSampleValue    = -32767.0;

SpeexCodec::SpeexCodec(const char * name, int mode, Direction dir)
  : H323FramedAudioCodec(name, dir)
{
  PTRACE(3, "Codec\tSpeex mode " << mode << " " << (dir == Encoder ? "en" : "de")
                << "coder created");

  bits = new SpeexBits;
  speex_bits_init(bits);

  if (direction == Encoder) {
      coder_state = speex_encoder_init(&speex_nb_mode);
      speex_encoder_ctl(coder_state, SPEEX_GET_FRAME_SIZE, &encoder_frame_size);
      speex_encoder_ctl(coder_state, SPEEX_SET_QUALITY,      &mode);
  } else {
      coder_state = speex_decoder_init(&speex_nb_mode);
  }
}

SpeexCodec::~SpeexCodec()
{
  speex_bits_destroy(bits);
  delete bits;

  if (direction == Encoder)
      speex_encoder_destroy(coder_state);
  else
      speex_decoder_destroy(coder_state);
}


BOOL SpeexCodec::EncodeFrame(BYTE * buffer, unsigned & length)
{
  // convert PCM to float
  float floatData[SAMPLES_PER_FRAME];
  PINDEX i;
  for (i = 0; i < SAMPLES_PER_FRAME; i++)
      floatData[i] = sampleBuffer[i];

  // encode PCM data in sampleBuffer to buffer
  speex_bits_reset(bits);
  speex_encode(coder_state, floatData, bits);

  length = speex_bits_write(bits, (char *)buffer, encoder_frame_size);

  return TRUE;
}


BOOL SpeexCodec::DecodeFrame(const BYTE * buffer, unsigned length, unsigned &)
{
  float floatData[SAMPLES_PER_FRAME];

  // decode Speex data to floats
  speex_bits_read_from(bits, (char *)buffer, length);
  speex_decode(coder_state, bits, floatData);

  // convert float to PCM
  PINDEX i;
  for (i = 0; i < SAMPLES_PER_FRAME; i++) {
      float sample = floatData[i];
      if (sample < MinSampleValue)
          sample = MinSampleValue;
      else if (sample > MaxSampleValue)
          sample = MaxSampleValue;
      sampleBuffer[i] = (short)sample;
  }

  return TRUE;
}



VC++ 中使用 API的 char 单字节压缩代码示例：

Encoding and decoding problem in speex 1.0.4
Subject:       Encoding and decoding problem in speex 1.0.4
List-id:       speex-dev.xiph.org

Hi,
                      I am using the speex 1.0.4 library from Windows.
                      I have posted my problem before but didn't get a solution. I am doing an
                      VOIP project
                      in which i am recording sound and streaming it to the peer. I wanted to
                      encode and decode
                      wav files that brought me to this site.
                      I am recording sound in the following format:-
                      m_WaveFormatEx.wFormatTag                  = WAVE_FORMAT_PCM;
                      m_WaveFormatEx.nChannels                    = 1;
                      m_WaveFormatEx.wBitsPerSample          = 8;
                      m_WaveFormatEx.cbSize                          = 0;
                      m_WaveFormatEx.nSamplesPerSec          = 8000;
                      m_WaveFormatEx.nBlockAlign                = 1;
                      m_WaveFormatEx.nAvgBytesPerSec        = 8000;
                      The recording is as follows :-
                      When the buffer(size = 2000 bytes) gets filled with sound data a
                      function with the body shown
                      below is called.
                      LPWAVEHDR lpHdr = (LPWAVEHDR) lParam;
                      if(lpHdr->dwBytesRecorded==0 || lpHdr==NULL)
                      return ERROR_SUCCESS;
                      ::waveInUnprepareHeader(m_hRecord, lpHdr, sizeof(WAVEHDR));
                      Here lpHdr->lpData contains the audio data in a character array.
                      Now here I want to use Speex codec for encoding the data so the encoding
                      function is
                      called (I am thankful to Tay YueWeng for the function).
                      char *encode(char *buffer, int &encodeSize)
                      {
                      char    *encodedBuffer = new char[RECBUFFER/2];                     
                      short    speexShort;
                      float  speexFloat[RECBUFFER/2];
                      void    *mEncode            = speex_encoder_init(&speex_nb_mode);
                     
                      speex_bits_init(&mBits);
                      // Convert the audio to a short then to a float buffer
                      int      halfBufferSize = RECBUFFER/2;
                      for (int i = 0; i < halfBufferSize; i++)
                      {
                      memcpy(&speexShort, &buffer[i*2], sizeof(short));
                      speexFloat[i]        = speexShort;
                      }
                      // Encode the sound data using the float buffer
                      speex_bits_reset(&mBits);
                      speex_encode(mEncode, speexFloat, &mBits);
                      encodeSize                      = speex_bits_write(&mBits, encodedBuffer,
                      RECBUFFER/2);
                     
                      speex_encoder_destroy(mEncode);
                     
                      speex_bits_destroy(&mBits);
                      // Return the encoded buffer
                      return encodedBuffer;
                      }
                      Here i noticed that though my captured audio data is 2000 bytes the
                      compressed form is
                      always 38 bytes. In the speexFloat array above i get values in the range
                      -32767 to +32767.
                      Is it correct. Also after calling the 'speex_encode' function the first
                      160 values in the
                      input float array i.e. speexFloat is changed (why does it happen?Is
                      anything abnormal).
                      Further after calling the above function for testing I decode the
                      returned encoded data
                      immediately by calling the decoding function shown bellow :-
                      char *decode (char *buffer, int encodeSize)
                      {
                      char *decodedBuffer    = new char[RECBUFFER];
                      short speexShort;
                      float speexFloat[RECBUFFER/2];
                      // Decode the sound data into a float buffer
                      void  *mDecode              = speex_decoder_init(&speex_nb_mode);
                     
                      speex_bits_init(&mBits);
                      int      halfBufferSize = RECBUFFER/2;
                      speex_bits_reset(&mBits);
                      speex_bits_read_from(&mBits, buffer, encodeSize);
                      speex_decode(mDecode, &mBits, speexFloat);
                      // Convert from float to short to char
                      for (int i = 0; i < halfBufferSize; i++)
                      {
                      speexShort = speexFloat[i];
                      memcpy(&decodedBuffer[i*2], &speexShort, sizeof(short));
                      }
                     
                      speex_encoder_destroy(mDecode);
                     
                      speex_bits_destroy(&mBits);
                      // Return the buffer
                      return decodedBuffer;
                      }
                      After decoding using the above function only the first 160 values in the
                      decodedBuffer array is
                      changed. i.e i encoded an 2000 byte audio data to get a 38 byte encoded
                      audio data. On decoding
                      the 38 byte audio data i get an decompressed 160 byte data. I don't
                      understand whats going
                      wrong. I checked all the messages posted in this newsgroup and did'nt
                      find an answer so i am
                      posting this code hoping that it gets solved soon.  Thanks in advance.