trunk/src/modules/audio_coding/main/source/acm_g7291.cc - vendor/opensource/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 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 "acm_g7291.h"
 #include "acm_common_defs.h"
 #include "acm_neteq.h"
 #include "trace.h"
 #include "webrtc_neteq.h"
 #include "webrtc_neteq_help_macros.h"

 #ifdef WEBRTC_CODEC_G729_1
     // NOTE! G.729.1 is not included in the open-source package. The following
     // interface file is needed:
     //
     // /modules/audio_coding/codecs/g7291/main/interface/g7291_interface.h
     //
     // The API in the header file should match the one below.
     //
     // int16_t WebRtcG7291_Create(G729_1_inst_t_** inst);
     // int16_t WebRtcG7291_Free(G729_1_inst_t_* inst);
     // int16_t WebRtcG7291_Encode(G729_1_inst_t_* encInst, int16_t* input,
     //                            int16_t* output, int16_t myRate,
     //                            int16_t nrFrames);
     // int16_t WebRtcG7291_EncoderInit(G729_1_inst_t_* encInst, int16_t myRate,
     //                                 int16_t flag8kHz, int16_t flagG729mode);
     // int16_t WebRtcG7291_Decode(G729_1_inst_t_* decInst);
     // int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
     // int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
     // int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
     // void WebRtcG7291_Version(char *versionStr, short len);
     #include "g7291_interface.h"
 #endif

 namespace webrtc {

 #ifndef WEBRTC_CODEC_G729_1

 ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
     : _encoderInstPtr(NULL),
       _decoderInstPtr(NULL),
       _myRate(32000),
       _flag8kHz(0),
       _flagG729mode(0) {
   return;
 }


 ACMG729_1::~ACMG729_1()
 {
     return;
 }


 WebRtc_Word16
 ACMG729_1::InternalEncode(
     WebRtc_UWord8* /* bitStream        */,
     WebRtc_Word16* /* bitStreamLenByte */)
 {
     return -1;
 }


 WebRtc_Word16
 ACMG729_1::DecodeSafe(
     WebRtc_UWord8* /* bitStream        */,
     WebRtc_Word16  /* bitStreamLenByte */,
     WebRtc_Word16* /* audio            */,
     WebRtc_Word16* /* audioSamples     */,
     WebRtc_Word8*  /* speechType       */)
 {
     return -1;
 }


 WebRtc_Word16
 ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* /* codecParams */)
 {
     return -1;
 }


 WebRtc_Word16
 ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codecParams */)
 {
     return -1;
 }


 WebRtc_Word32
 ACMG729_1::CodecDef(
     WebRtcNetEQ_CodecDef& /* codecDef  */,
     const CodecInst&      /* codecInst */)
 {
     return -1;
 }


 ACMGenericCodec*
 ACMG729_1::CreateInstance(void)
 {
     return NULL;
 }


 WebRtc_Word16
 ACMG729_1::InternalCreateEncoder()
 {
     return -1;
 }


 void
 ACMG729_1::DestructEncoderSafe()
 {
     return;
 }


 WebRtc_Word16
 ACMG729_1::InternalCreateDecoder()
 {
     return -1;
 }


 void
 ACMG729_1::DestructDecoderSafe()
 {
     return;
 }


 void
 ACMG729_1::InternalDestructEncoderInst(
     void* /* ptrInst */)
 {
     return;
 }


 WebRtc_Word16
 ACMG729_1::UnregisterFromNetEqSafe(
     ACMNetEQ*     /* netEq       */,
     WebRtc_Word16 /* payloadType */)
 {
     return -1;
 }

 WebRtc_Word16
 ACMG729_1::SetBitRateSafe(
     const WebRtc_Word32 /*rate*/ )
 {
   return -1;
 }

 #else     //===================== Actual Implementation =======================

 struct G729_1_inst_t_;

 ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
     : _encoderInstPtr(NULL),
       _decoderInstPtr(NULL),
       _myRate(32000),  // Default rate.
       _flag8kHz(0),
       _flagG729mode(0) {
   // TODO(tlegrand): We should add codecID as a input variable to the
   // constructor of ACMGenericCodec.
   _codecID = codecID;
   return;
 }

 ACMG729_1::~ACMG729_1()
 {
     if(_encoderInstPtr != NULL)
     {
         WebRtcG7291_Free(_encoderInstPtr);
         _encoderInstPtr = NULL;
     }
     if(_decoderInstPtr != NULL)
     {
         WebRtcG7291_Free(_decoderInstPtr);
         _decoderInstPtr = NULL;
     }
     return;
 }


 WebRtc_Word16
 ACMG729_1::InternalEncode(
     WebRtc_UWord8* bitStream,
     WebRtc_Word16* bitStreamLenByte)
 {

     // Initialize before entering the loop
     WebRtc_Word16 noEncodedSamples = 0;
     *bitStreamLenByte = 0;

   WebRtc_Word16 byteLengthFrame = 0;

     // Derive number of 20ms frames per encoded packet.
   // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
     WebRtc_Word16 n20msFrames = (_frameLenSmpl / 320);
     // Byte length for the frame. +1 is for rate information.
     byteLengthFrame = _myRate/(8*50) * n20msFrames + (1 - _flagG729mode);

     // The following might be revised if we have G729.1 Annex C (support for DTX);
     do
     {
         *bitStreamLenByte = WebRtcG7291_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
        (WebRtc_Word16*)bitStream, _myRate, n20msFrames);

         // increment the read index this tell the caller that how far
         // we have gone forward in reading the audio buffer
     _inAudioIxRead += 160;

         // sanity check
         if(*bitStreamLenByte < 0)
         {
       // error has happened
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
         "InternalEncode: Encode error for G729_1");
             *bitStreamLenByte = 0;
             return -1;
         }

     noEncodedSamples += 160;
     } while(*bitStreamLenByte == 0);


     // This criteria will change if we have Annex C.
     if(*bitStreamLenByte != byteLengthFrame)
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
             "InternalEncode: Encode error for G729_1");
         *bitStreamLenByte = 0;
         return -1;
     }


     if(noEncodedSamples != _frameLenSmpl)
     {
         *bitStreamLenByte = 0;
         return -1;
     }

     return *bitStreamLenByte;
 }


 WebRtc_Word16
 ACMG729_1::DecodeSafe(
     WebRtc_UWord8* /* bitStream        */,
     WebRtc_Word16  /* bitStreamLenByte */,
     WebRtc_Word16* /* audio            */,
     WebRtc_Word16* /* audioSamples     */,
     WebRtc_Word8*  /* speechType       */)
 {
     return 0;
 }


 WebRtc_Word16
 ACMG729_1::InternalInitEncoder(
     WebRtcACMCodecParams* codecParams)
 {
   //set the bit rate and initialize
   _myRate = codecParams->codecInstant.rate;
     return SetBitRateSafe( (WebRtc_UWord32)_myRate);
 }


 WebRtc_Word16
 ACMG729_1::InternalInitDecoder(
     WebRtcACMCodecParams* /* codecParams */)
 {
     if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0)
   {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
         "InternalInitDecoder: init decoder failed for G729_1");
     return -1;
   }
   return 0;
 }


 WebRtc_Word32
 ACMG729_1::CodecDef(
     WebRtcNetEQ_CodecDef& codecDef,
     const CodecInst&      codecInst)
 {
     if (!_decoderInitialized)
     {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
       "CodeDef: Decoder uninitialized for G729_1");
       return -1;
     }

     // Fill up the structure by calling
     // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
     // Then call NetEQ to add the codec to it's
     // database.
     SET_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype,
         _decoderInstPtr, 16000);
     SET_G729_1_FUNCTIONS((codecDef));
     return 0;
 }


 ACMGenericCodec*
 ACMG729_1::CreateInstance(void)
 {
     return NULL;
 }


 WebRtc_Word16
 ACMG729_1::InternalCreateEncoder()
 {
     if (WebRtcG7291_Create(&_encoderInstPtr) < 0)
   {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
       "InternalCreateEncoder: create encoder failed for G729_1");
     return -1;
   }
   return 0;
 }


 void
 ACMG729_1::DestructEncoderSafe()
 {
     _encoderExist = false;
     _encoderInitialized = false;
     if(_encoderInstPtr != NULL)
     {
         WebRtcG7291_Free(_encoderInstPtr);
         _encoderInstPtr = NULL;
     }
 }


 WebRtc_Word16
 ACMG729_1::InternalCreateDecoder()
 {
    if (WebRtcG7291_Create(&_decoderInstPtr) < 0)
    {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
     "InternalCreateDecoder: create decoder failed for G729_1");
      return -1;
    }
    return 0;
 }


 void
 ACMG729_1::DestructDecoderSafe()
 {
     _decoderExist = false;
     _decoderInitialized = false;
     if(_decoderInstPtr != NULL)
     {
         WebRtcG7291_Free(_decoderInstPtr);
         _decoderInstPtr = NULL;
     }
 }


 void
 ACMG729_1::InternalDestructEncoderInst(
     void* ptrInst)
 {
     if(ptrInst != NULL)
     {
         //WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
     }
     return;
 }


 WebRtc_Word16
 ACMG729_1::UnregisterFromNetEqSafe(
     ACMNetEQ*     netEq,
     WebRtc_Word16 payloadType)
 {
     if(payloadType != _decoderParams.codecInstant.pltype)
     {
         WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
       "Cannot unregister codec: given payload-type does not match \
 the stored payload type",
             _decoderParams.codecInstant.plname,
             payloadType,
             _decoderParams.codecInstant.pltype);
         return -1;
     }
     return netEq->RemoveCodec(kDecoderG729_1);
 }

 WebRtc_Word16
 ACMG729_1::SetBitRateSafe(
     const WebRtc_Word32 rate)
 {
     //allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
     //                22000, 24000, 26000, 28000, 30000, 32000};
     // TODO(tlegrand): This check exists in one other place two. Should be
     // possible to reuse code.
     switch(rate)
     {
     case 8000:
         {
             _myRate = 8000;
             break;
         }
   case 12000:
         {
             _myRate = 12000;
             break;
         }
   case 14000:
         {
             _myRate = 14000;
             break;
         }
   case 16000:
         {
             _myRate = 16000;
             break;
         }
   case 18000:
         {
             _myRate = 18000;
             break;
         }
   case 20000:
         {
             _myRate = 20000;
             break;
         }
   case 22000:
         {
             _myRate = 22000;
             break;
         }
   case 24000:
         {
             _myRate = 24000;
             break;
         }
   case 26000:
         {
             _myRate = 26000;
             break;
         }
   case 28000:
         {
             _myRate = 28000;
             break;
         }
   case 30000:
         {
             _myRate = 30000;
             break;
         }
   case 32000:
         {
             _myRate = 32000;
             break;
         }
     default:
         {
       WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
         "SetBitRateSafe: Invalid rate G729_1");
             return -1;
         }
     }

     // Re-init with new rate
     if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz, _flagG729mode) >= 0)
     {
         _encoderParams.codecInstant.rate = _myRate;
         return 0;
     }
     else
     {
         return -1;
     }
 }


 #endif

 } // namespace webrtc
	/*
	* Copyright (c) 2012 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 "acm_g7291.h"
	#include "acm_common_defs.h"
	#include "acm_neteq.h"
	#include "trace.h"
	#include "webrtc_neteq.h"
	#include "webrtc_neteq_help_macros.h"

	#ifdef WEBRTC_CODEC_G729_1
	// NOTE! G.729.1 is not included in the open-source package. The following
	// interface file is needed:
	//
	// /modules/audio_coding/codecs/g7291/main/interface/g7291_interface.h
	//
	// The API in the header file should match the one below.
	//
	// int16_t WebRtcG7291_Create(G729_1_inst_t_** inst);
	// int16_t WebRtcG7291_Free(G729_1_inst_t_* inst);
	// int16_t WebRtcG7291_Encode(G729_1_inst_t_* encInst, int16_t* input,
	// int16_t* output, int16_t myRate,
	// int16_t nrFrames);
	// int16_t WebRtcG7291_EncoderInit(G729_1_inst_t_* encInst, int16_t myRate,
	// int16_t flag8kHz, int16_t flagG729mode);
	// int16_t WebRtcG7291_Decode(G729_1_inst_t_* decInst);
	// int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
	// int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
	// int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
	// void WebRtcG7291_Version(char *versionStr, short len);
	#include "g7291_interface.h"
	#endif

	namespace webrtc {

	#ifndef WEBRTC_CODEC_G729_1

	ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
	: _encoderInstPtr(NULL),
	_decoderInstPtr(NULL),
	_myRate(32000),
	_flag8kHz(0),
	_flagG729mode(0) {
	return;
	}


	ACMG729_1::~ACMG729_1()
	{
	return;
	}


	WebRtc_Word16
	ACMG729_1::InternalEncode(
	WebRtc_UWord8* /* bitStream */,
	WebRtc_Word16* /* bitStreamLenByte */)
	{
	return -1;
	}


	WebRtc_Word16
	ACMG729_1::DecodeSafe(
	WebRtc_UWord8* /* bitStream */,
	WebRtc_Word16 /* bitStreamLenByte */,
	WebRtc_Word16* /* audio */,
	WebRtc_Word16* /* audioSamples */,
	WebRtc_Word8* /* speechType */)
	{
	return -1;
	}


	WebRtc_Word16
	ACMG729_1::InternalInitEncoder(
	WebRtcACMCodecParams* /* codecParams */)
	{
	return -1;
	}


	WebRtc_Word16
	ACMG729_1::InternalInitDecoder(
	WebRtcACMCodecParams* /* codecParams */)
	{
	return -1;
	}


	WebRtc_Word32
	ACMG729_1::CodecDef(
	WebRtcNetEQ_CodecDef& /* codecDef */,
	const CodecInst& /* codecInst */)
	{
	return -1;
	}


	ACMGenericCodec*
	ACMG729_1::CreateInstance(void)
	{
	return NULL;
	}


	WebRtc_Word16
	ACMG729_1::InternalCreateEncoder()
	{
	return -1;
	}


	void
	ACMG729_1::DestructEncoderSafe()
	{
	return;
	}


	WebRtc_Word16
	ACMG729_1::InternalCreateDecoder()
	{
	return -1;
	}


	void
	ACMG729_1::DestructDecoderSafe()
	{
	return;
	}


	void
	ACMG729_1::InternalDestructEncoderInst(
	void* /* ptrInst */)
	{
	return;
	}


	WebRtc_Word16
	ACMG729_1::UnregisterFromNetEqSafe(
	ACMNetEQ* /* netEq */,
	WebRtc_Word16 /* payloadType */)
	{
	return -1;
	}

	WebRtc_Word16
	ACMG729_1::SetBitRateSafe(
	const WebRtc_Word32 /rate/ )
	{
	return -1;
	}

	#else //===================== Actual Implementation =======================

	struct G729_1_inst_t_;

	ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
	: _encoderInstPtr(NULL),
	_decoderInstPtr(NULL),
	_myRate(32000), // Default rate.
	_flag8kHz(0),
	_flagG729mode(0) {
	// TODO(tlegrand): We should add codecID as a input variable to the
	// constructor of ACMGenericCodec.
	_codecID = codecID;
	return;
	}

	ACMG729_1::~ACMG729_1()
	{
	if(_encoderInstPtr != NULL)
	{
	WebRtcG7291_Free(_encoderInstPtr);
	_encoderInstPtr = NULL;
	}
	if(_decoderInstPtr != NULL)
	{
	WebRtcG7291_Free(_decoderInstPtr);
	_decoderInstPtr = NULL;
	}
	return;
	}


	WebRtc_Word16
	ACMG729_1::InternalEncode(
	WebRtc_UWord8* bitStream,
	WebRtc_Word16* bitStreamLenByte)
	{

	// Initialize before entering the loop
	WebRtc_Word16 noEncodedSamples = 0;
	*bitStreamLenByte = 0;

	WebRtc_Word16 byteLengthFrame = 0;

	// Derive number of 20ms frames per encoded packet.
	// [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
	WebRtc_Word16 n20msFrames = (_frameLenSmpl / 320);
	// Byte length for the frame. +1 is for rate information.
	byteLengthFrame = _myRate/(850) n20msFrames + (1 - _flagG729mode);

	// The following might be revised if we have G729.1 Annex C (support for DTX);
	do
	{
	*bitStreamLenByte = WebRtcG7291_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
	(WebRtc_Word16*)bitStream, _myRate, n20msFrames);

	// increment the read index this tell the caller that how far
	// we have gone forward in reading the audio buffer
	_inAudioIxRead += 160;

	// sanity check
	if(*bitStreamLenByte < 0)
	{
	// error has happened
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"InternalEncode: Encode error for G729_1");
	*bitStreamLenByte = 0;
	return -1;
	}

	noEncodedSamples += 160;
	} while(*bitStreamLenByte == 0);


	// This criteria will change if we have Annex C.
	if(*bitStreamLenByte != byteLengthFrame)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"InternalEncode: Encode error for G729_1");
	*bitStreamLenByte = 0;
	return -1;
	}


	if(noEncodedSamples != _frameLenSmpl)
	{
	*bitStreamLenByte = 0;
	return -1;
	}

	return *bitStreamLenByte;
	}


	WebRtc_Word16
	ACMG729_1::DecodeSafe(
	WebRtc_UWord8* /* bitStream */,
	WebRtc_Word16 /* bitStreamLenByte */,
	WebRtc_Word16* /* audio */,
	WebRtc_Word16* /* audioSamples */,
	WebRtc_Word8* /* speechType */)
	{
	return 0;
	}


	WebRtc_Word16
	ACMG729_1::InternalInitEncoder(
	WebRtcACMCodecParams* codecParams)
	{
	//set the bit rate and initialize
	_myRate = codecParams->codecInstant.rate;
	return SetBitRateSafe( (WebRtc_UWord32)_myRate);
	}


	WebRtc_Word16
	ACMG729_1::InternalInitDecoder(
	WebRtcACMCodecParams* /* codecParams */)
	{
	if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"InternalInitDecoder: init decoder failed for G729_1");
	return -1;
	}
	return 0;
	}


	WebRtc_Word32
	ACMG729_1::CodecDef(
	WebRtcNetEQ_CodecDef& codecDef,
	const CodecInst& codecInst)
	{
	if (!_decoderInitialized)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"CodeDef: Decoder uninitialized for G729_1");
	return -1;
	}

	// Fill up the structure by calling
	// "SET_CODEC_PAR" & "SET_G729_FUNCTION."
	// Then call NetEQ to add the codec to it's
	// database.
	SET_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype,
	_decoderInstPtr, 16000);
	SET_G729_1_FUNCTIONS((codecDef));
	return 0;
	}


	ACMGenericCodec*
	ACMG729_1::CreateInstance(void)
	{
	return NULL;
	}


	WebRtc_Word16
	ACMG729_1::InternalCreateEncoder()
	{
	if (WebRtcG7291_Create(&_encoderInstPtr) < 0)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"InternalCreateEncoder: create encoder failed for G729_1");
	return -1;
	}
	return 0;
	}


	void
	ACMG729_1::DestructEncoderSafe()
	{
	_encoderExist = false;
	_encoderInitialized = false;
	if(_encoderInstPtr != NULL)
	{
	WebRtcG7291_Free(_encoderInstPtr);
	_encoderInstPtr = NULL;
	}
	}


	WebRtc_Word16
	ACMG729_1::InternalCreateDecoder()
	{
	if (WebRtcG7291_Create(&_decoderInstPtr) < 0)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"InternalCreateDecoder: create decoder failed for G729_1");
	return -1;
	}
	return 0;
	}


	void
	ACMG729_1::DestructDecoderSafe()
	{
	_decoderExist = false;
	_decoderInitialized = false;
	if(_decoderInstPtr != NULL)
	{
	WebRtcG7291_Free(_decoderInstPtr);
	_decoderInstPtr = NULL;
	}
	}


	void
	ACMG729_1::InternalDestructEncoderInst(
	void* ptrInst)
	{
	if(ptrInst != NULL)
	{
	//WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
	}
	return;
	}


	WebRtc_Word16
	ACMG729_1::UnregisterFromNetEqSafe(
	ACMNetEQ* netEq,
	WebRtc_Word16 payloadType)
	{
	if(payloadType != _decoderParams.codecInstant.pltype)
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"Cannot unregister codec: given payload-type does not match \
	the stored payload type",
	_decoderParams.codecInstant.plname,
	payloadType,
	_decoderParams.codecInstant.pltype);
	return -1;
	}
	return netEq->RemoveCodec(kDecoderG729_1);
	}

	WebRtc_Word16
	ACMG729_1::SetBitRateSafe(
	const WebRtc_Word32 rate)
	{
	//allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
	// 22000, 24000, 26000, 28000, 30000, 32000};
	// TODO(tlegrand): This check exists in one other place two. Should be
	// possible to reuse code.
	switch(rate)
	{
	case 8000:
	{
	_myRate = 8000;
	break;
	}
	case 12000:
	{
	_myRate = 12000;
	break;
	}
	case 14000:
	{
	_myRate = 14000;
	break;
	}
	case 16000:
	{
	_myRate = 16000;
	break;
	}
	case 18000:
	{
	_myRate = 18000;
	break;
	}
	case 20000:
	{
	_myRate = 20000;
	break;
	}
	case 22000:
	{
	_myRate = 22000;
	break;
	}
	case 24000:
	{
	_myRate = 24000;
	break;
	}
	case 26000:
	{
	_myRate = 26000;
	break;
	}
	case 28000:
	{
	_myRate = 28000;
	break;
	}
	case 30000:
	{
	_myRate = 30000;
	break;
	}
	case 32000:
	{
	_myRate = 32000;
	break;
	}
	default:
	{
	WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
	"SetBitRateSafe: Invalid rate G729_1");
	return -1;
	}
	}

	// Re-init with new rate
	if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz, _flagG729mode) >= 0)
	{
	_encoderParams.codecInstant.rate = _myRate;
	return 0;
	}
	else
	{
	return -1;
	}
	}


	#endif

	} // namespace webrtc