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Reformating files in audio coding module.

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This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org 2012-11-05 09:35:51 +00:00
parent a56d759723
commit f7fa6276e2
26 changed files with 5715 additions and 7532 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

329
webrtc/modules/audio_coding/main/source/acm_g729.cc
View File

@ -16,25 +16,10 @@
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_G729
// NOTE! G.729 is not included in the open-source package. The following
// interface file is needed:
//
// /modules/audio_coding/codecs/g729/main/interface/g729_interface.h
//
// The API in the header file should match the one below.
//
// int16_t WebRtcG729_CreateEnc(G729_encinst_t_** inst);
// int16_t WebRtcG729_CreateDec(G729_decinst_t_** inst);
// int16_t WebRtcG729_FreeEnc(G729_encinst_t_* inst);
// int16_t WebRtcG729_FreeDec(G729_decinst_t_* inst);
// int16_t WebRtcG729_Encode(G729_encinst_t_* encInst, int16_t* input,
// int16_t len, int16_t* output);
// int16_t WebRtcG729_EncoderInit(G729_encinst_t_* encInst, int16_t mode);
// int16_t WebRtcG729_Decode(G729_decinst_t_* decInst);
// int16_t WebRtcG729_DecodeBwe(G729_decinst_t_* decInst, int16_t* input);
// int16_t WebRtcG729_DecodePlc(G729_decinst_t_* decInst);
// int16_t WebRtcG729_DecoderInit(G729_decinst_t_* decInst);
#include "g729_interface.h"
// NOTE! G.729 is not included in the open-source package. Modify this file
// or your codec API to match the function calls and names of used G.729 API
// file.
#include "g729_interface.h"
#endif
namespace webrtc {
@ -47,152 +32,96 @@ ACMG729::ACMG729(WebRtc_Word16 /* codecID */)
return;
}
ACMG729::~ACMG729()
{
ACMG729::~ACMG729() {
return;
}
WebRtc_Word16
ACMG729::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMG729::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
return -1;
}
WebRtc_Word16
ACMG729::EnableDTX()
{
WebRtc_Word16 ACMG729::EnableDTX() {
return -1;
}
WebRtc_Word16
ACMG729::DisableDTX()
{
WebRtc_Word16 ACMG729::DisableDTX() {
return -1;
}
WebRtc_Word32
ACMG729::ReplaceInternalDTXSafe(
const bool /*replaceInternalDTX*/)
{
WebRtc_Word32 ACMG729::ReplaceInternalDTXSafe(
const bool /*replaceInternalDTX*/) {
return -1;
}
WebRtc_Word32
ACMG729::IsInternalDTXReplacedSafe(
bool* /* internalDTXReplaced */)
{
WebRtc_Word32 ACMG729::IsInternalDTXReplacedSafe(
bool* /* internalDTXReplaced */) {
return -1;
}
WebRtc_Word16
ACMG729::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMG729::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return -1;
}
WebRtc_Word16
ACMG729::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMG729::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word16
ACMG729::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMG729::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word32
ACMG729::CodecDef(
WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */)
{
WebRtc_Word32 ACMG729::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */) {
return -1;
}
ACMGenericCodec*
ACMG729::CreateInstance(void)
{
ACMGenericCodec* ACMG729::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMG729::InternalCreateEncoder()
{
WebRtc_Word16 ACMG729::InternalCreateEncoder() {
return -1;
}
void
ACMG729::DestructEncoderSafe()
{
void ACMG729::DestructEncoderSafe() {
return;
}
WebRtc_Word16
ACMG729::InternalCreateDecoder()
{
WebRtc_Word16 ACMG729::InternalCreateDecoder() {
return -1;
}
void
ACMG729::DestructDecoderSafe()
{
void ACMG729::DestructDecoderSafe() {
return;
}
void
ACMG729::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMG729::InternalDestructEncoderInst(void* /* ptrInst */) {
return;
}
#else //===================== Actual Implementation =======================
ACMG729::ACMG729(
WebRtc_Word16 codecID):
_encoderInstPtr(NULL),
_decoderInstPtr(NULL)
{
ACMG729::ACMG729(WebRtc_Word16 codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL) {
_codecID = codecID;
_hasInternalDTX = true;
return;
}
ACMG729::~ACMG729()
{
if(_encoderInstPtr != NULL)
{
ACMG729::~ACMG729() {
if (_encoderInstPtr != NULL) {
// Delete encoder memory
WebRtcG729_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
// Delete decoder memory
WebRtcG729_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
@ -200,32 +129,26 @@ ACMG729::~ACMG729()
return;
}
WebRtc_Word16
ACMG729::InternalEncode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte)
{
WebRtc_Word16 ACMG729::InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) {
// Initialize before entering the loop
WebRtc_Word16 noEncodedSamples = 0;
WebRtc_Word16 tmpLenByte = 0;
WebRtc_Word16 vadDecision = 0;
*bitStreamLenByte = 0;
while(noEncodedSamples < _frameLenSmpl)
{
while (noEncodedSamples < _frameLenSmpl) {
// Call G.729 encoder with pointer to encoder memory, input
// audio, number of samples and bitsream
tmpLenByte = WebRtcG729_Encode(_encoderInstPtr,
&_inAudio[_inAudioIxRead], 80,
(WebRtc_Word16*)(&(bitStream[*bitStreamLenByte])));
tmpLenByte = WebRtcG729_Encode(
_encoderInstPtr, &_inAudio[_inAudioIxRead], 80,
(WebRtc_Word16*) (&(bitStream[*bitStreamLenByte])));
// increment the read index this tell the caller that how far
// we have gone forward in reading the audio buffer
_inAudioIxRead += 80;
// sanity check
if(tmpLenByte < 0)
{
if (tmpLenByte < 0) {
// error has happened
*bitStreamLenByte = 0;
return -1;
@ -233,12 +156,9 @@ ACMG729::InternalEncode(
// increment number of written bytes
*bitStreamLenByte += tmpLenByte;
switch(tmpLenByte)
{
case 0:
{
if(0 == noEncodedSamples)
{
switch (tmpLenByte) {
case 0: {
if (0 == noEncodedSamples) {
// this is the first 10 ms in this packet and there is
// no data generated, perhaps DTX is enabled and the
// codec is not generating any bit-stream for this 10 ms.
@ -247,14 +167,11 @@ ACMG729::InternalEncode(
}
break;
}
case 2:
{
case 2: {
// check if G.729 internal DTX is enabled
if(_hasInternalDTX && _dtxEnabled)
{
if (_hasInternalDTX && _dtxEnabled) {
vadDecision = 0;
for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
{
for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
_vadLabel[n] = vadDecision;
}
}
@ -262,14 +179,12 @@ ACMG729::InternalEncode(
// how much audio we have encoded
return *bitStreamLenByte;
}
case 10:
{
case 10: {
vadDecision = 1;
// this is a valid length just continue encoding
break;
}
default:
{
default: {
return -1;
}
}
@ -279,10 +194,8 @@ ACMG729::InternalEncode(
}
// update VAD decision vector
if(_hasInternalDTX && !vadDecision && _dtxEnabled)
{
for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
{
if (_hasInternalDTX && !vadDecision && _dtxEnabled) {
for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
_vadLabel[n] = vadDecision;
}
}
@ -291,78 +204,52 @@ ACMG729::InternalEncode(
return *bitStreamLenByte;
}
WebRtc_Word16
ACMG729::EnableDTX()
{
if(_dtxEnabled)
{
WebRtc_Word16 ACMG729::EnableDTX() {
if (_dtxEnabled) {
// DTX already enabled, do nothing
return 0;
}
else if(_encoderExist)
{
} else if (_encoderExist) {
// Re-init the G.729 encoder to turn on DTX
if(WebRtcG729_EncoderInit(_encoderInstPtr, 1) < 0)
{
if (WebRtcG729_EncoderInit(_encoderInstPtr, 1) < 0) {
return -1;
}
_dtxEnabled = true;
return 0;
}
else
{
} else {
return -1;
}
}
WebRtc_Word16
ACMG729::DisableDTX()
{
if(!_dtxEnabled)
{
WebRtc_Word16 ACMG729::DisableDTX() {
if (!_dtxEnabled) {
// DTX already dissabled, do nothing
return 0;
}
else if(_encoderExist)
{
} else if (_encoderExist) {
// Re-init the G.729 decoder to turn off DTX
if(WebRtcG729_EncoderInit(_encoderInstPtr, 0) < 0)
{
if (WebRtcG729_EncoderInit(_encoderInstPtr, 0) < 0) {
return -1;
}
_dtxEnabled = false;
return 0;
}
else
{
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}
}
WebRtc_Word32
ACMG729::ReplaceInternalDTXSafe(
const bool replaceInternalDTX)
{
// This function is used to dissable the G.729 built in DTX and use an
WebRtc_Word32 ACMG729::ReplaceInternalDTXSafe(const bool replaceInternalDTX) {
// This function is used to disable the G.729 built in DTX and use an
// external instead.
if(replaceInternalDTX == _hasInternalDTX)
{
if (replaceInternalDTX == _hasInternalDTX) {
// Make sure we keep the DTX/VAD setting if possible
bool oldEnableDTX = _dtxEnabled;
bool oldEnableVAD = _vadEnabled;
ACMVADMode oldMode = _vadMode;
if (replaceInternalDTX)
{
if (replaceInternalDTX) {
// Disable internal DTX before enabling external DTX
DisableDTX();
}
else
{
} else {
// Disable external DTX before enabling internal
ACMGenericCodec::DisableDTX();
}
@ -381,56 +268,36 @@ ACMG729::ReplaceInternalDTXSafe(
return 0;
}
WebRtc_Word32
ACMG729::IsInternalDTXReplacedSafe(
bool* internalDTXReplaced)
{
WebRtc_Word32 ACMG729::IsInternalDTXReplacedSafe(bool* internalDTXReplaced) {
// Get status of wether DTX is replaced or not
*internalDTXReplaced = !_hasInternalDTX;
return 0;
}
WebRtc_Word16
ACMG729::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMG729::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
// This function is not used. G.729 decoder is called from inside NetEQ
return 0;
}
WebRtc_Word16
ACMG729::InternalInitEncoder(
WebRtcACMCodecParams* codecParams)
{
WebRtc_Word16 ACMG729::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
// Init G.729 encoder
return WebRtcG729_EncoderInit(_encoderInstPtr,
((codecParams->enableDTX)? 1:0));
((codecParams->enableDTX) ? 1 : 0));
}
WebRtc_Word16
ACMG729::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMG729::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
// Init G.729 decoder
return WebRtcG729_DecoderInit(_decoderInstPtr);
}
WebRtc_Word32
ACMG729::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
WebRtc_Word32 ACMG729::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) {
if (!_decoderInitialized) {
// Todo:
// log error
return -1;
@ -440,72 +307,50 @@ ACMG729::CodecDef(
// "SET_CODEC_PAR" & "SET_G729_FUNCTION."
// Then call NetEQ to add the codec to it's
// database.
SET_CODEC_PAR((codecDef), kDecoderG729, codecInst.pltype,
_decoderInstPtr, 8000);
SET_CODEC_PAR((codecDef), kDecoderG729, codecInst.pltype, _decoderInstPtr,
8000);
SET_G729_FUNCTIONS((codecDef));
return 0;
}
ACMGenericCodec*
ACMG729::CreateInstance(void)
{
ACMGenericCodec* ACMG729::CreateInstance(void) {
// Function not used
return NULL;
}
WebRtc_Word16
ACMG729::InternalCreateEncoder()
{
WebRtc_Word16 ACMG729::InternalCreateEncoder() {
// Create encoder memory
return WebRtcG729_CreateEnc(&_encoderInstPtr);
}
void
ACMG729::DestructEncoderSafe()
{
void ACMG729::DestructEncoderSafe() {
// Free encoder memory
_encoderExist = false;
_encoderInitialized = false;
if(_encoderInstPtr != NULL)
{
if (_encoderInstPtr != NULL) {
WebRtcG729_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
}
WebRtc_Word16
ACMG729::InternalCreateDecoder()
{
WebRtc_Word16 ACMG729::InternalCreateDecoder() {
// Create decoder memory
return WebRtcG729_CreateDec(&_decoderInstPtr);
}
void
ACMG729::DestructDecoderSafe()
{
void ACMG729::DestructDecoderSafe() {
// Free decoder memory
_decoderExist = false;
_decoderInitialized = false;
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcG729_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
}
}
void
ACMG729::InternalDestructEncoderInst(
void* ptrInst)
{
if(ptrInst != NULL)
{
WebRtcG729_FreeEnc((G729_encinst_t_*)ptrInst);
void ACMG729::InternalDestructEncoderInst(void* ptrInst) {
if (ptrInst != NULL) {
WebRtcG729_FreeEnc((G729_encinst_t_*) ptrInst);
}
return;
}

31
webrtc/modules/audio_coding/main/source/acm_g729.h
View File

@ -19,34 +19,28 @@ struct G729_decinst_t_;
namespace webrtc {
class ACMG729 : public ACMGenericCodec
{
public:
class ACMG729 : public ACMGenericCodec {
public:
ACMG729(WebRtc_Word16 codecID);
~ACMG729();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -57,18 +51,15 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 EnableDTX();
WebRtc_Word16 DisableDTX();
WebRtc_Word32 ReplaceInternalDTXSafe(
const bool replaceInternalDTX);
WebRtc_Word32 ReplaceInternalDTXSafe(const bool replaceInternalDTX);
WebRtc_Word32 IsInternalDTXReplacedSafe(
bool* internalDTXReplaced);
WebRtc_Word32 IsInternalDTXReplacedSafe(bool* internalDTXReplaced);
G729_encinst_t_* _encoderInstPtr;
G729_decinst_t_* _decoderInstPtr;

286
webrtc/modules/audio_coding/main/source/acm_g7291.cc
View File

@ -16,24 +16,9 @@
#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);
// NOTE! G.729.1 is not included in the open-source package. Modify this file
// or your codec API to match the function calls and names of used G.729.1 API
// file.
#include "g7291_interface.h"
#endif
@ -41,7 +26,7 @@ namespace webrtc {
#ifndef WEBRTC_CODEC_G729_1
ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
ACMG729_1::ACMG729_1(WebRtc_Word16 /* codecID */)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL),
_myRate(32000),
@ -50,105 +35,63 @@ ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
return;
}
ACMG729_1::~ACMG729_1()
{
ACMG729_1::~ACMG729_1() {
return;
}
WebRtc_Word16
ACMG729_1::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMG729_1::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
return -1;
}
WebRtc_Word16
ACMG729_1::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return -1;
}
WebRtc_Word16
ACMG729_1::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMG729_1::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word16
ACMG729_1::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMG729_1::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word32
ACMG729_1::CodecDef(
WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */)
{
WebRtc_Word32 ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */) {
return -1;
}
ACMGenericCodec*
ACMG729_1::CreateInstance(void)
{
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMG729_1::InternalCreateEncoder()
{
WebRtc_Word16 ACMG729_1::InternalCreateEncoder() {
return -1;
}
void
ACMG729_1::DestructEncoderSafe()
{
void ACMG729_1::DestructEncoderSafe() {
return;
}
WebRtc_Word16
ACMG729_1::InternalCreateDecoder()
{
WebRtc_Word16 ACMG729_1::InternalCreateDecoder() {
return -1;
}
void
ACMG729_1::DestructDecoderSafe()
{
void ACMG729_1::DestructDecoderSafe() {
return;
}
void
ACMG729_1::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMG729_1::InternalDestructEncoderInst(void* /* ptrInst */) {
return;
}
WebRtc_Word16
ACMG729_1::SetBitRateSafe(
const WebRtc_Word32 /*rate*/ )
{
WebRtc_Word16 ACMG729_1::SetBitRateSafe(const WebRtc_Word32 /*rate*/) {
return -1;
}
@ -168,27 +111,20 @@ ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
return;
}
ACMG729_1::~ACMG729_1()
{
if(_encoderInstPtr != NULL)
{
ACMG729_1::~ACMG729_1() {
if (_encoderInstPtr != NULL) {
WebRtcG7291_Free(_encoderInstPtr);
_encoderInstPtr = NULL;
}
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcG7291_Free(_decoderInstPtr);
_decoderInstPtr = NULL;
}
return;
}
WebRtc_Word16
ACMG729_1::InternalEncode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte)
{
WebRtc_Word16 ACMG729_1::InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) {
// Initialize before entering the loop
WebRtc_Word16 noEncodedSamples = 0;
@ -200,21 +136,21 @@ ACMG729_1::InternalEncode(
// [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);
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);
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)
{
if (*bitStreamLenByte < 0) {
// error has happened
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalEncode: Encode error for G729_1");
@ -223,21 +159,17 @@ ACMG729_1::InternalEncode(
}
noEncodedSamples += 160;
} while(*bitStreamLenByte == 0);
} while (*bitStreamLenByte == 0);
// This criteria will change if we have Annex C.
if(*bitStreamLenByte != byteLengthFrame)
{
if (*bitStreamLenByte != byteLengthFrame) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalEncode: Encode error for G729_1");
*bitStreamLenByte = 0;
return -1;
}
if(noEncodedSamples != _frameLenSmpl)
{
if (noEncodedSamples != _frameLenSmpl) {
*bitStreamLenByte = 0;
return -1;
}
@ -245,35 +177,24 @@ ACMG729_1::InternalEncode(
return *bitStreamLenByte;
}
WebRtc_Word16
ACMG729_1::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return 0;
}
WebRtc_Word16
ACMG729_1::InternalInitEncoder(
WebRtcACMCodecParams* codecParams)
{
WebRtc_Word16 ACMG729_1::InternalInitEncoder(
WebRtcACMCodecParams* codecParams) {
//set the bit rate and initialize
_myRate = codecParams->codecInstant.rate;
return SetBitRateSafe( (WebRtc_UWord32)_myRate);
return SetBitRateSafe((WebRtc_UWord32) _myRate);
}
WebRtc_Word16
ACMG729_1::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0)
{
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;
@ -281,14 +202,9 @@ ACMG729_1::InternalInitDecoder(
return 0;
}
WebRtc_Word32
ACMG729_1::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
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;
@ -298,25 +214,18 @@ ACMG729_1::CodecDef(
// "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_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype, _decoderInstPtr,
16000);
SET_G729_1_FUNCTIONS((codecDef));
return 0;
}
ACMGenericCodec*
ACMG729_1::CreateInstance(void)
{
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMG729_1::InternalCreateEncoder()
{
if (WebRtcG7291_Create(&_encoderInstPtr) < 0)
{
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;
@ -324,25 +233,17 @@ ACMG729_1::InternalCreateEncoder()
return 0;
}
void
ACMG729_1::DestructEncoderSafe()
{
void ACMG729_1::DestructEncoderSafe() {
_encoderExist = false;
_encoderInitialized = false;
if(_encoderInstPtr != NULL)
{
if (_encoderInstPtr != NULL) {
WebRtcG7291_Free(_encoderInstPtr);
_encoderInstPtr = NULL;
}
}
WebRtc_Word16
ACMG729_1::InternalCreateDecoder()
{
if (WebRtcG7291_Create(&_decoderInstPtr) < 0)
{
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;
@ -350,103 +251,77 @@ ACMG729_1::InternalCreateDecoder()
return 0;
}
void
ACMG729_1::DestructDecoderSafe()
{
void ACMG729_1::DestructDecoderSafe() {
_decoderExist = false;
_decoderInitialized = false;
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcG7291_Free(_decoderInstPtr);
_decoderInstPtr = NULL;
}
}
void
ACMG729_1::InternalDestructEncoderInst(
void* ptrInst)
{
if(ptrInst != NULL)
{
void ACMG729_1::InternalDestructEncoderInst(void* ptrInst) {
if (ptrInst != NULL) {
//WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
}
return;
}
WebRtc_Word16
ACMG729_1::SetBitRateSafe(
const WebRtc_Word32 rate)
{
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:
{
switch (rate) {
case 8000: {
_myRate = 8000;
break;
}
case 12000:
{
case 12000: {
_myRate = 12000;
break;
}
case 14000:
{
case 14000: {
_myRate = 14000;
break;
}
case 16000:
{
case 16000: {
_myRate = 16000;
break;
}
case 18000:
{
case 18000: {
_myRate = 18000;
break;
}
case 20000:
{
case 20000: {
_myRate = 20000;
break;
}
case 22000:
{
case 22000: {
_myRate = 22000;
break;
}
case 24000:
{
case 24000: {
_myRate = 24000;
break;
}
case 26000:
{
case 26000: {
_myRate = 26000;
break;
}
case 28000:
{
case 28000: {
_myRate = 28000;
break;
}
case 30000:
{
case 30000: {
_myRate = 30000;
break;
}
case 32000:
{
case 32000: {
_myRate = 32000;
break;
}
default:
{
default: {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"SetBitRateSafe: Invalid rate G729_1");
return -1;
@ -454,18 +329,15 @@ ACMG729_1::SetBitRateSafe(
}
// Re-init with new rate
if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz, _flagG729mode) >= 0)
{
if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz,
_flagG729mode) >= 0) {
_encoderParams.codecInstant.rate = _myRate;
return 0;
}
else
{
} else {
return -1;
}
}
#endif
} // namespace webrtc

29
webrtc/modules/audio_coding/main/source/acm_g7291.h
View File

@ -19,34 +19,28 @@ struct G729_1_inst_t_;
namespace webrtc {
class ACMG729_1: public ACMGenericCodec
{
public:
class ACMG729_1 : public ACMGenericCodec {
public:
ACMG729_1(WebRtc_Word16 codecID);
~ACMG729_1();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -57,11 +51,9 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 SetBitRateSafe(
const WebRtc_Word32 rate);
WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
G729_1_inst_t_* _encoderInstPtr;
G729_1_inst_t_* _decoderInstPtr;
@ -69,7 +61,6 @@ protected:
WebRtc_UWord16 _myRate;
WebRtc_Word16 _flag8kHz;
WebRtc_Word16 _flagG729mode;
};
} // namespace webrtc

862
webrtc/modules/audio_coding/main/source/acm_generic_codec.cc
View File

File diff suppressed because it is too large Load Diff

235
webrtc/modules/audio_coding/main/source/acm_generic_codec.h
View File

@ -23,35 +23,30 @@
struct WebRtcVadInst;
struct WebRtcCngEncInst;
namespace webrtc
{
namespace webrtc {
// forward declaration
struct CodecInst;
class ACMNetEQ;
class ACMGenericCodec
{
public:
class ACMGenericCodec {
public:
///////////////////////////////////////////////////////////////////////////
// Constructor of the class
//
ACMGenericCodec();
///////////////////////////////////////////////////////////////////////////
// Destructor of the class.
//
virtual ~ACMGenericCodec();
///////////////////////////////////////////////////////////////////////////
// ACMGenericCodec* CreateInstance();
// The function will be used for FEC. It is not implemented yet.
//
virtual ACMGenericCodec* CreateInstance() = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 Encode()
// The function is called to perform an encoding of the audio stored in
@ -105,13 +100,11 @@ public:
// -1 if error is occurred, otherwise the length of the bit-stream in
// bytes.
//
WebRtc_Word16 Encode(
WebRtc_UWord8* bitStream,
WebRtc_Word16 Encode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte,
WebRtc_UWord32* timeStamp,
WebRtcACMEncodingType* encodingType);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 Decode()
// This function is used to decode a given bit-stream, without engaging
@ -134,11 +127,8 @@ public:
// -1 if failed to decode,
// 0 if succeeded.
//
WebRtc_Word16 Decode(
WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word16 Decode(WebRtc_UWord8* bitStream, WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
///////////////////////////////////////////////////////////////////////////
@ -157,7 +147,8 @@ public:
// we simply copy the data and return it both for
// left channel and right channel decoding.
virtual void SplitStereoPacket(WebRtc_UWord8* /* payload */,
WebRtc_Word32* /* payload_length */) {}
WebRtc_Word32* /* payload_length */) {
}
///////////////////////////////////////////////////////////////////////////
// bool EncoderInitialized();
@ -168,7 +159,6 @@ public:
//
bool EncoderInitialized();
///////////////////////////////////////////////////////////////////////////
// bool DecoderInitialized();
//
@ -178,7 +168,6 @@ public:
//
bool DecoderInitialized();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 EncoderParams()
// It is called to get encoder parameters. It will call
@ -194,9 +183,7 @@ public:
// 0 otherwise.
//
//
WebRtc_Word16 EncoderParams(
WebRtcACMCodecParams *encParams);
WebRtc_Word16 EncoderParams(WebRtcACMCodecParams *encParams);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 DecoderParams(...)
@ -213,11 +200,9 @@ public:
// 0 otherwise.
//
//
bool DecoderParams(
WebRtcACMCodecParams *decParams,
bool DecoderParams(WebRtcACMCodecParams *decParams,
const WebRtc_UWord8 payloadType);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InitEncoder(...)
// This function is called to initialize the encoder with the given
@ -234,11 +219,9 @@ public:
// -1 if failed to initialize.
//
//
WebRtc_Word16 InitEncoder(
WebRtcACMCodecParams* codecParams,
WebRtc_Word16 InitEncoder(WebRtcACMCodecParams* codecParams,
bool forceInitialization);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InitDecoder()
// This function is called to initialize the decoder with the given
@ -256,11 +239,9 @@ public:
// -1 if failed to initialize.
//
//
WebRtc_Word16 InitDecoder(
WebRtcACMCodecParams* codecParams,
WebRtc_Word16 InitDecoder(WebRtcACMCodecParams* codecParams,
bool forceInitialization);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 RegisterInNetEq(...)
// This function is called to register the decoder in NetEq, with the given
@ -274,10 +255,7 @@ public:
// -1 if failed to register,
// 0 if successfully initialized.
//
WebRtc_Word32 RegisterInNetEq(
ACMNetEQ* netEq,
const CodecInst& codecInst);
WebRtc_Word32 RegisterInNetEq(ACMNetEQ* netEq, const CodecInst& codecInst);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 Add10MsData(...)
@ -298,13 +276,11 @@ public:
// -1 if failed
// 0 otherwise.
//
WebRtc_Word32 Add10MsData(
const WebRtc_UWord32 timeStamp,
WebRtc_Word32 Add10MsData(const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* data,
const WebRtc_UWord16 length,
const WebRtc_UWord8 audioChannel);
///////////////////////////////////////////////////////////////////////////
// WebRtc_UWord32 NoMissedSamples()
// This function returns the number of samples which are overwritten in
@ -317,7 +293,6 @@ public:
//
WebRtc_UWord32 NoMissedSamples() const;
///////////////////////////////////////////////////////////////////////////
// void ResetNoMissedSamples()
// This function resets the number of overwritten samples to zero.
@ -325,7 +300,6 @@ public:
//
void ResetNoMissedSamples();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 SetBitRate()
// The function is called to set the encoding rate.
@ -340,7 +314,6 @@ public:
//
WebRtc_Word16 SetBitRate(const WebRtc_Word32 bitRateBPS);
///////////////////////////////////////////////////////////////////////////
// DestructEncoderInst()
// This API is used in conferencing. It will free the memory that is pointed
@ -351,8 +324,7 @@ public:
// -ptrInst : pointer to an encoder instance to be deleted.
//
//
void DestructEncoderInst(
void* ptrInst);
void DestructEncoderInst(void* ptrInst);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 AudioBuffer()
@ -369,9 +341,7 @@ public:
// -1 if fails to copy the audio buffer,
// 0 if succeeded.
//
WebRtc_Word16 AudioBuffer(
WebRtcACMAudioBuff& audioBuff);
WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audioBuff);
///////////////////////////////////////////////////////////////////////////
// WebRtc_UWord32 EarliestTimestamp()
@ -383,7 +353,6 @@ public:
//
WebRtc_UWord32 EarliestTimestamp() const;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 SetAudioBuffer()
// This function is called to set the audio buffer and the associated
@ -395,8 +364,6 @@ public:
//
WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 SetVAD()
// This is called to set VAD & DTX. If the codec has internal DTX that will
@ -424,11 +391,9 @@ public:
// -1 if failed to set DTX & VAD as specified,
// 0 if succeeded.
//
WebRtc_Word16 SetVAD(
const bool enableDTX = true,
const bool enableVAD = false,
const ACMVADMode mode = VADNormal);
WebRtc_Word16 SetVAD(const bool enableDTX = true,
const bool enableVAD = false, const ACMVADMode mode =
VADNormal);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ReplaceInternalDTX()
@ -447,15 +412,15 @@ public:
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 IsInternalDTXReplaced()
// This is called to check if the codec internal DTX is replaced by WebRtc DTX.
// This is only valid for G729 where the user has possibility to replace
// This is called to check if the codec internal DTX is replaced by WebRtc
// DTX. This is only valid for G729 where the user has possibility to replace
// AnnexB with WebRtc DTX. For other codecs this function has no effect.
//
// Output:
// -internalDTXReplaced : if true the internal DTX is replaced with WebRtc.
//
// Return value
// -1 if failed to check if replace internal DTX or replacement not feasible,
// -1 if failed to check
// 0 if succeeded.
//
WebRtc_Word32 IsInternalDTXReplaced(bool* internalDTXReplaced);
@ -467,13 +432,10 @@ public:
// Input:
// -netEqDecodeLock : pointer to the lock associated with NetEQ of ACM.
//
void SetNetEqDecodeLock(
RWLockWrapper* netEqDecodeLock)
{
void SetNetEqDecodeLock(RWLockWrapper* netEqDecodeLock) {
_netEqDecodeLock = netEqDecodeLock;
}
///////////////////////////////////////////////////////////////////////////
// bool HasInternalDTX()
// Used to check if the codec has internal DTX.
@ -482,12 +444,10 @@ public:
// true if the codec has an internal DTX, e.g. G729,
// false otherwise.
//
bool HasInternalDTX() const
{
bool HasInternalDTX() const {
return _hasInternalDTX;
}
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 GetEstimatedBandwidth()
// Used to get decoder estimated bandwidth. Only iSAC will provide a value.
@ -499,7 +459,6 @@ public:
//
WebRtc_Word32 GetEstimatedBandwidth();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 SetEstimatedBandwidth()
// Used to set estiamted bandwidth sent out of band from other side. Only
@ -527,11 +486,9 @@ public:
// -1 if fails to get codec specific RED,
// 0 if succeeded.
//
WebRtc_Word32 GetRedPayload(
WebRtc_UWord8* redPayload,
WebRtc_Word32 GetRedPayload(WebRtc_UWord8* redPayload,
WebRtc_Word16* payloadBytes);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 ResetEncoder()
// By calling this function you would re-initialize the encoder with the
@ -544,7 +501,6 @@ public:
//
WebRtc_Word16 ResetEncoder();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 ResetEncoder()
// By calling this function you would re-initialize the decoder with the
@ -554,9 +510,7 @@ public:
// -1 if failed,
// 0 if succeeded.
//
WebRtc_Word16 ResetDecoder(
WebRtc_Word16 payloadType);
WebRtc_Word16 ResetDecoder(WebRtc_Word16 payloadType);
///////////////////////////////////////////////////////////////////////////
// void DestructEncoder()
@ -567,7 +521,6 @@ public:
//
void DestructEncoder();
///////////////////////////////////////////////////////////////////////////
// void DestructDecoder()
// This function is called to delete the decoder instance, if possible, to
@ -578,7 +531,6 @@ public:
//
void DestructDecoder();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 SamplesLeftToEncode()
// Returns the number of samples required to be able to do encoding.
@ -588,7 +540,6 @@ public:
//
WebRtc_Word16 SamplesLeftToEncode();
///////////////////////////////////////////////////////////////////////////
// WebRtc_UWord32 LastEncodedTimestamp()
// Returns the timestamp of the last frame it encoded.
@ -598,7 +549,6 @@ public:
//
WebRtc_UWord32 LastEncodedTimestamp() const;
///////////////////////////////////////////////////////////////////////////
// SetUniqueID()
// Set a unique ID for the codec to be used for tracing and debuging
@ -606,9 +556,7 @@ public:
// Input
// -id : A number to identify the codec.
//
void SetUniqueID(
const WebRtc_UWord32 id);
void SetUniqueID(const WebRtc_UWord32 id);
///////////////////////////////////////////////////////////////////////////
// IsAudioBufferFresh()
@ -620,7 +568,6 @@ public:
//
bool IsAudioBufferFresh() const;
///////////////////////////////////////////////////////////////////////////
// UpdateDecoderSampFreq()
// For most of the codecs this function does nothing. It must be
@ -646,13 +593,10 @@ public:
// 0 if succeeded in updating the decoder.
// -1 if failed to update.
//
virtual WebRtc_Word16 UpdateDecoderSampFreq(
WebRtc_Word16 /* codecId */)
{
virtual WebRtc_Word16 UpdateDecoderSampFreq(WebRtc_Word16 /* codecId */) {
return 0;
}
///////////////////////////////////////////////////////////////////////////
// UpdateEncoderSampFreq()
// Call this function to update the encoder sampling frequency. This
@ -669,9 +613,7 @@ public:
// -1 if failed, or if this is meaningless for the given codec.
// 0 if succeeded.
//
virtual WebRtc_Word16 UpdateEncoderSampFreq(
WebRtc_UWord16 encoderSampFreqHz);
virtual WebRtc_Word16 UpdateEncoderSampFreq(WebRtc_UWord16 encoderSampFreqHz);
///////////////////////////////////////////////////////////////////////////
// EncoderSampFreq()
@ -685,9 +627,7 @@ public:
// -1 if failed to output sampling rate.
// 0 if the sample rate is returned successfully.
//
virtual WebRtc_Word16 EncoderSampFreq(
WebRtc_UWord16& sampFreqHz);
virtual WebRtc_Word16 EncoderSampFreq(WebRtc_UWord16& sampFreqHz);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word32 ConfigISACBandwidthEstimator()
@ -715,9 +655,7 @@ public:
//
virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
const WebRtc_UWord8 initFrameSizeMsec,
const WebRtc_UWord16 initRateBitPerSec,
const bool enforceFrameSize);
const WebRtc_UWord16 initRateBitPerSec, const bool enforceFrameSize);
///////////////////////////////////////////////////////////////////////////
// SetISACMaxPayloadSize()
@ -737,7 +675,6 @@ public:
virtual WebRtc_Word32 SetISACMaxPayloadSize(
const WebRtc_UWord16 maxPayloadLenBytes);
///////////////////////////////////////////////////////////////////////////
// SetISACMaxRate()
// Set the maximum instantaneous rate of iSAC. For a payload of B bits
@ -753,9 +690,7 @@ public:
// -1 if failed to set the maximum rate.
// 0 if the maximum rate is set successfully.
//
virtual WebRtc_Word32 SetISACMaxRate(
const WebRtc_UWord32 maxRateBitPerSec);
virtual WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 maxRateBitPerSec);
///////////////////////////////////////////////////////////////////////////
// SaveDecoderParamS()
@ -765,20 +700,14 @@ public:
// -codecParams : pointer to a struct where the parameters of
// decoder is stored in.
//
void SaveDecoderParam(
const WebRtcACMCodecParams* codecParams);
void SaveDecoderParam(const WebRtcACMCodecParams* codecParams);
WebRtc_Word32 FrameSize()
{
WebRtc_Word32 FrameSize() {
return _frameLenSmpl;
}
void SetIsMaster(bool isMaster);
///////////////////////////////////////////////////////////////////////////
// REDPayloadISAC()
// This is an iSAC-specific function. The function is called to get RED
@ -803,8 +732,7 @@ public:
// is returned.
//
//
virtual WebRtc_Word16 REDPayloadISAC(
const WebRtc_Word32 isacRate,
virtual WebRtc_Word16 REDPayloadISAC(const WebRtc_Word32 isacRate,
const WebRtc_Word16 isacBwEstimate,
WebRtc_UWord8* payload,
WebRtc_Word16* payloadLenBytes);
@ -821,7 +749,7 @@ public:
return false;
}
protected:
protected:
///////////////////////////////////////////////////////////////////////////
// All the functions with FunctionNameSafe(...) contain the actual
// implementation of FunctionName(...). FunctionName() acquires an
@ -834,8 +762,7 @@ protected:
// See Encode() for the description of function, input(s)/output(s) and
// return value.
//
WebRtc_Word16 EncodeSafe(
WebRtc_UWord8* bitStream,
WebRtc_Word16 EncodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte,
WebRtc_UWord32* timeStamp,
WebRtcACMEncodingType* encodingType);
@ -844,8 +771,7 @@ protected:
// See Decode() for the description of function, input(s)/output(s) and
// return value.
//
virtual WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
virtual WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
@ -855,8 +781,7 @@ protected:
// See Add10MsSafe() for the description of function, input(s)/output(s)
// and return value.
//
virtual WebRtc_Word32 Add10MsDataSafe(
const WebRtc_UWord32 timeStamp,
virtual WebRtc_Word32 Add10MsDataSafe(const WebRtc_UWord32 timeStamp,
const WebRtc_Word16* data,
const WebRtc_UWord16 length,
const WebRtc_UWord8 audioChannel);
@ -865,16 +790,14 @@ protected:
// See RegisterInNetEq() for the description of function,
// input(s)/output(s) and return value.
//
virtual WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
virtual WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) = 0;
///////////////////////////////////////////////////////////////////////////
// See EncoderParam() for the description of function, input(s)/output(s)
// and return value.
//
WebRtc_Word16 EncoderParamsSafe(
WebRtcACMCodecParams *encParams);
WebRtc_Word16 EncoderParamsSafe(WebRtcACMCodecParams *encParams);
///////////////////////////////////////////////////////////////////////////
// See DecoderParam for the description of function, input(s)/output(s)
@ -890,8 +813,7 @@ protected:
// can be registered as receive codec. Hence two payloads are associated
// with a single codec instance.
//
virtual bool DecoderParamsSafe(
WebRtcACMCodecParams *decParams,
virtual bool DecoderParamsSafe(WebRtcACMCodecParams *decParams,
const WebRtc_UWord8 payloadType);
///////////////////////////////////////////////////////////////////////////
@ -904,24 +826,21 @@ protected:
// See InitEncoder() for the description of function, input(s)/output(s)
// and return value.
//
WebRtc_Word16 InitEncoderSafe(
WebRtcACMCodecParams *codecParams,
WebRtc_Word16 InitEncoderSafe(WebRtcACMCodecParams *codecParams,
bool forceInitialization);
///////////////////////////////////////////////////////////////////////////
// See InitDecoder() for the description of function, input(s)/output(s)
// and return value.
//
WebRtc_Word16 InitDecoderSafe(
WebRtcACMCodecParams *codecParams,
WebRtc_Word16 InitDecoderSafe(WebRtcACMCodecParams *codecParams,
bool forceInitialization);
///////////////////////////////////////////////////////////////////////////
// See ResetDecoder() for the description of function, input(s)/output(s)
// and return value.
//
WebRtc_Word16 ResetDecoderSafe(
WebRtc_Word16 payloadType);
WebRtc_Word16 ResetDecoderSafe(WebRtc_Word16 payloadType);
///////////////////////////////////////////////////////////////////////////
// See DestructEncoder() for the description of function,
@ -941,51 +860,47 @@ protected:
//
// Any codec that can change the bit-rate has to implement this.
//
virtual WebRtc_Word16 SetBitRateSafe(
const WebRtc_Word32 bitRateBPS);
virtual WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 bitRateBPS);
///////////////////////////////////////////////////////////////////////////
// See GetEstimatedBandwidth() for the description of function, input(s)/output(s)
// and return value.
// See GetEstimatedBandwidth() for the description of function,
// input(s)/output(s) and return value.
//
virtual WebRtc_Word32 GetEstimatedBandwidthSafe();
///////////////////////////////////////////////////////////////////////////
// See SetEstimatedBandwidth() for the description of function, input(s)/output(s)
// and return value.
// See SetEstimatedBandwidth() for the description of function,
// input(s)/output(s) and return value.
//
virtual WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
virtual WebRtc_Word32 SetEstimatedBandwidthSafe(
WebRtc_Word32 estimatedBandwidth);
///////////////////////////////////////////////////////////////////////////
// See GetRedPayload() for the description of function, input(s)/output(s)
// and return value.
//
virtual WebRtc_Word32 GetRedPayloadSafe(
WebRtc_UWord8* redPayload,
virtual WebRtc_Word32 GetRedPayloadSafe(WebRtc_UWord8* redPayload,
WebRtc_Word16* payloadBytes);
///////////////////////////////////////////////////////////////////////////
// See SetVAD() for the description of function, input(s)/output(s) and
// return value.
//
WebRtc_Word16 SetVADSafe(
const bool enableDTX = true,
const bool enableVAD = false,
WebRtc_Word16 SetVADSafe(const bool enableDTX = true, const bool enableVAD =
false,
const ACMVADMode mode = VADNormal);
///////////////////////////////////////////////////////////////////////////
// See ReplaceInternalDTX() for the description of function, input and
// return value.
//
virtual WebRtc_Word32 ReplaceInternalDTXSafe(
const bool replaceInternalDTX);
virtual WebRtc_Word32 ReplaceInternalDTXSafe(const bool replaceInternalDTX);
///////////////////////////////////////////////////////////////////////////
// See IsInternalDTXReplaced() for the description of function, input and
// return value.
//
virtual WebRtc_Word32 IsInternalDTXReplacedSafe(
bool* internalDTXReplaced);
virtual WebRtc_Word32 IsInternalDTXReplacedSafe(bool* internalDTXReplaced);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 CreateEncoder()
@ -997,7 +912,6 @@ protected:
//
WebRtc_Word16 CreateEncoder();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 CreateDecoder()
// Creates the decoder instance.
@ -1008,7 +922,6 @@ protected:
//
WebRtc_Word16 CreateDecoder();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 EnableVAD();
// Enables VAD with the given mode. The VAD instance will be created if
@ -1024,7 +937,6 @@ protected:
//
WebRtc_Word16 EnableVAD(ACMVADMode mode);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 DisableVAD()
// Disables VAD.
@ -1035,7 +947,6 @@ protected:
//
WebRtc_Word16 DisableVAD();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 EnableDTX()
// Enables DTX. This method should be overwritten for codecs which have
@ -1047,7 +958,6 @@ protected:
//
virtual WebRtc_Word16 EnableDTX();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 DisableDTX()
// Disables usage of DTX. This method should be overwritten for codecs which
@ -1059,7 +969,6 @@ protected:
//
virtual WebRtc_Word16 DisableDTX();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalEncode()
// This is a codec-specific function called in EncodeSafe() to actually
@ -1075,11 +984,9 @@ protected:
// -1 if failed,
// otherwise the length of the bit-stream is returned.
//
virtual WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitStream,
virtual WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalInitEncoder()
// This is a codec-specific function called in InitEncoderSafe(), it has to
@ -1102,7 +1009,6 @@ protected:
virtual WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalInitDecoder()
// This is a codec-specific function called in InitDecoderSafe(), it has to
@ -1120,7 +1026,6 @@ protected:
virtual WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams) = 0;
///////////////////////////////////////////////////////////////////////////
// void IncreaseNoMissedSamples()
// This method is called to increase the number of samples that are
@ -1130,9 +1035,7 @@ protected:
// -noSamples : the number of overwritten samples is incremented
// by this value.
//
void IncreaseNoMissedSamples(
const WebRtc_Word16 noSamples);
void IncreaseNoMissedSamples(const WebRtc_Word16 noSamples);
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalCreateEncoder()
@ -1146,7 +1049,6 @@ protected:
//
virtual WebRtc_Word16 InternalCreateEncoder() = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalCreateDecoder()
// This is a codec-specific method called in CreateDecoderSafe() it is
@ -1159,7 +1061,6 @@ protected:
//
virtual WebRtc_Word16 InternalCreateDecoder() = 0;
///////////////////////////////////////////////////////////////////////////
// void InternalDestructEncoderInst()
// This is a codec-specific method, used in conferencing, called from
@ -1175,9 +1076,7 @@ protected:
// -1 if failed,
// 0 if succeeded.
//
virtual void InternalDestructEncoderInst(
void* ptrInst) = 0;
virtual void InternalDestructEncoderInst(void* ptrInst) = 0;
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 InternalResetEncoder()
@ -1194,8 +1093,6 @@ protected:
//
virtual WebRtc_Word16 InternalResetEncoder();
///////////////////////////////////////////////////////////////////////////
// WebRtc_Word16 ProcessFrameVADDTX()
// This function is called when a full frame of audio is available. It will
@ -1226,12 +1123,10 @@ protected:
// -1 if failed,
// 0 if succeeded.
//
WebRtc_Word16 ProcessFrameVADDTX(
WebRtc_UWord8* bitStream,
WebRtc_Word16 ProcessFrameVADDTX(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte,
WebRtc_Word16* samplesProcessed);
///////////////////////////////////////////////////////////////////////////
// CanChangeEncodingParam()
// Check if the codec parameters can be changed. In conferencing normally
@ -1243,7 +1138,6 @@ protected:
//
virtual bool CanChangeEncodingParam(CodecInst& codecInst);
///////////////////////////////////////////////////////////////////////////
// CurrentRate()
// Call to get the current encoding rate of the encoder. This function
@ -1254,14 +1148,11 @@ protected:
// Output:
// -rateBitPerSec : the current target rate of the codec.
//
virtual void CurrentRate(
WebRtc_Word32& /* rateBitPerSec */)
{
virtual void CurrentRate(WebRtc_Word32& /* rateBitPerSec */) {
return;
}
virtual void SaveDecoderParamSafe(
const WebRtcACMCodecParams* codecParams);
virtual void SaveDecoderParamSafe(const WebRtcACMCodecParams* codecParams);
// &_inAudio[_inAudioIxWrite] always point to where new audio can be
// written to

265
webrtc/modules/audio_coding/main/source/acm_gsmfr.cc
View File

@ -16,25 +16,10 @@
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_GSMFR
// NOTE! GSM-FR is not included in the open-source package. The following
// interface file is needed:
//
// /modules/audio_coding/codecs/gsmfr/main/interface/gsmfr_interface.h
//
// The API in the header file should match the one below.
//
// int16_t WebRtcGSMFR_CreateEnc(GSMFR_encinst_t_** inst);
// int16_t WebRtcGSMFR_CreateDec(GSMFR_decinst_t_** inst);
// int16_t WebRtcGSMFR_FreeEnc(GSMFR_encinst_t_* inst);
// int16_t WebRtcGSMFR_FreeDec(GSMFR_decinst_t_* inst);
// int16_t WebRtcGSMFR_Encode(GSMFR_encinst_t_* encInst, int16_t* input,
// int16_t len, int16_t* output);
// int16_t WebRtcGSMFR_EncoderInit(GSMFR_encinst_t_* encInst, int16_t mode);
// int16_t WebRtcGSMFR_Decode(GSMFR_decinst_t_* decInst);
// int16_t WebRtcGSMFR_DecodeBwe(GSMFR_decinst_t_* decInst, int16_t* input);
// int16_t WebRtcGSMFR_DecodePlc(GSMFR_decinst_t_* decInst);
// int16_t WebRtcGSMFR_DecoderInit(GSMFR_decinst_t_* decInst);
#include "gsmfr_interface.h"
// NOTE! GSM-FR is not included in the open-source package. Modify this file
// or your codec API to match the function calls and names of used GSM-FR API
// file.
#include "gsmfr_interface.h"
#endif
namespace webrtc {
@ -47,240 +32,157 @@ ACMGSMFR::ACMGSMFR(WebRtc_Word16 /* codecID */)
return;
}
ACMGSMFR::~ACMGSMFR()
{
ACMGSMFR::~ACMGSMFR() {
return;
}
WebRtc_Word16
ACMGSMFR::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMGSMFR::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
return -1;
}
WebRtc_Word16
ACMGSMFR::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return -1;
}
WebRtc_Word16
ACMGSMFR::EnableDTX()
{
WebRtc_Word16 ACMGSMFR::EnableDTX() {
return -1;
}
WebRtc_Word16
ACMGSMFR::DisableDTX()
{
WebRtc_Word16 ACMGSMFR::DisableDTX() {
return -1;
}
WebRtc_Word16
ACMGSMFR::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMGSMFR::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word16
ACMGSMFR::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word32
ACMGSMFR::CodecDef(
WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */)
{
WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */) {
return -1;
}
ACMGenericCodec*
ACMGSMFR::CreateInstance(void)
{
ACMGenericCodec* ACMGSMFR::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMGSMFR::InternalCreateEncoder()
{
WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
return -1;
}
void
ACMGSMFR::DestructEncoderSafe()
{
void ACMGSMFR::DestructEncoderSafe() {
return;
}
WebRtc_Word16
ACMGSMFR::InternalCreateDecoder()
{
WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
return -1;
}
void
ACMGSMFR::DestructDecoderSafe()
{
void ACMGSMFR::DestructDecoderSafe() {
return;
}
void
ACMGSMFR::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMGSMFR::InternalDestructEncoderInst(void* /* ptrInst */) {
return;
}
#else //===================== Actual Implementation =======================
ACMGSMFR::ACMGSMFR(
WebRtc_Word16 codecID):
_encoderInstPtr(NULL),
_decoderInstPtr(NULL)
{
ACMGSMFR::ACMGSMFR(WebRtc_Word16 codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL) {
_codecID = codecID;
_hasInternalDTX = true;
return;
}
ACMGSMFR::~ACMGSMFR()
{
if(_encoderInstPtr != NULL)
{
ACMGSMFR::~ACMGSMFR() {
if (_encoderInstPtr != NULL) {
WebRtcGSMFR_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcGSMFR_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
}
return;
}
WebRtc_Word16
ACMGSMFR::InternalEncode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte)
{
WebRtc_Word16 ACMGSMFR::InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) {
*bitStreamLenByte = WebRtcGSMFR_Encode(_encoderInstPtr,
&_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
&_inAudio[_inAudioIxRead],
_frameLenSmpl,
(WebRtc_Word16*) bitStream);
// increment the read index this tell the caller that how far
// we have gone forward in reading the audio buffer
_inAudioIxRead += _frameLenSmpl;
return *bitStreamLenByte;
}
WebRtc_Word16
ACMGSMFR::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return 0;
}
WebRtc_Word16
ACMGSMFR::EnableDTX()
{
if(_dtxEnabled)
{
WebRtc_Word16 ACMGSMFR::EnableDTX() {
if (_dtxEnabled) {
return 0;
}
else if(_encoderExist)
{
if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 1) < 0)
{
} else if (_encoderExist) {
if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, 1) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"EnableDTX: cannot init encoder for GSMFR");
return -1;
}
_dtxEnabled = true;
return 0;
}
else
{
} else {
return -1;
}
}
WebRtc_Word16
ACMGSMFR::DisableDTX()
{
if(!_dtxEnabled)
{
WebRtc_Word16 ACMGSMFR::DisableDTX() {
if (!_dtxEnabled) {
return 0;
}
else if(_encoderExist)
{
if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 0) < 0)
{
} else if (_encoderExist) {
if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, 0) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"DisableDTX: cannot init encoder for GSMFR");
return -1;
}
_dtxEnabled = false;
return 0;
}
else
{
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}
}
WebRtc_Word16
ACMGSMFR::InternalInitEncoder(
WebRtcACMCodecParams* codecParams)
{
if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, ((codecParams->enableDTX)? 1:0)) < 0)
{
WebRtc_Word16 ACMGSMFR::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
if (WebRtcGSMFR_EncoderInit(_encoderInstPtr,
((codecParams->enableDTX) ? 1 : 0)) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalInitEncoder: cannot init encoder for GSMFR");
}
return 0;
}
WebRtc_Word16
ACMGSMFR::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
if (WebRtcGSMFR_DecoderInit(_decoderInstPtr) < 0)
{
WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
if (WebRtcGSMFR_DecoderInit(_decoderInstPtr) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalInitDecoder: cannot init decoder for GSMFR");
return -1;
@ -288,14 +190,9 @@ ACMGSMFR::InternalInitDecoder(
return 0;
}
WebRtc_Word32
ACMGSMFR::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) {
if (!_decoderInitialized) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"CodecDef: decoder is not initialized for GSMFR");
return -1;
@ -304,25 +201,18 @@ ACMGSMFR::CodecDef(
// "SET_CODEC_PAR" & "SET_GSMFR_FUNCTION."
// Then call NetEQ to add the codec to it's
// database.
SET_CODEC_PAR((codecDef), kDecoderGSMFR, codecInst.pltype,
_decoderInstPtr, 8000);
SET_CODEC_PAR((codecDef), kDecoderGSMFR, codecInst.pltype, _decoderInstPtr,
8000);
SET_GSMFR_FUNCTIONS((codecDef));
return 0;
}
ACMGenericCodec*
ACMGSMFR::CreateInstance(void)
{
ACMGenericCodec* ACMGSMFR::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMGSMFR::InternalCreateEncoder()
{
if (WebRtcGSMFR_CreateEnc(&_encoderInstPtr) < 0)
{
WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
if (WebRtcGSMFR_CreateEnc(&_encoderInstPtr) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalCreateEncoder: cannot create instance for GSMFR encoder");
return -1;
@ -330,12 +220,8 @@ ACMGSMFR::InternalCreateEncoder()
return 0;
}
void
ACMGSMFR::DestructEncoderSafe()
{
if(_encoderInstPtr != NULL)
{
void ACMGSMFR::DestructEncoderSafe() {
if (_encoderInstPtr != NULL) {
WebRtcGSMFR_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
@ -343,12 +229,8 @@ ACMGSMFR::DestructEncoderSafe()
_encoderInitialized = false;
}
WebRtc_Word16
ACMGSMFR::InternalCreateDecoder()
{
if (WebRtcGSMFR_CreateDec(&_decoderInstPtr) < 0)
{
WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
if (WebRtcGSMFR_CreateDec(&_decoderInstPtr) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalCreateDecoder: cannot create instance for GSMFR decoder");
return -1;
@ -356,12 +238,8 @@ ACMGSMFR::InternalCreateDecoder()
return 0;
}
void
ACMGSMFR::DestructDecoderSafe()
{
if(_decoderInstPtr != NULL)
{
void ACMGSMFR::DestructDecoderSafe() {
if (_decoderInstPtr != NULL) {
WebRtcGSMFR_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
}
@ -369,14 +247,9 @@ ACMGSMFR::DestructDecoderSafe()
_decoderInitialized = false;
}
void
ACMGSMFR::InternalDestructEncoderInst(
void* ptrInst)
{
if(ptrInst != NULL)
{
WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*)ptrInst);
void ACMGSMFR::InternalDestructEncoderInst(void* ptrInst) {
if (ptrInst != NULL) {
WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*) ptrInst);
}
return;
}

25
webrtc/modules/audio_coding/main/source/acm_gsmfr.h
View File

@ -19,34 +19,28 @@ struct GSMFR_decinst_t_;
namespace webrtc {
class ACMGSMFR : public ACMGenericCodec
{
public:
class ACMGSMFR : public ACMGenericCodec {
public:
ACMGSMFR(WebRtc_Word16 codecID);
~ACMGSMFR();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -57,8 +51,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 EnableDTX();

244
webrtc/modules/audio_coding/main/source/acm_ilbc.cc
View File

@ -16,11 +16,10 @@
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_ILBC
#include "ilbc.h"
#include "ilbc.h"
#endif
namespace webrtc
{
namespace webrtc {
#ifndef WEBRTC_CODEC_ILBC
@ -30,145 +29,94 @@ ACMILBC::ACMILBC(WebRtc_Word16 /* codecID */)
return;
}
ACMILBC::~ACMILBC()
{
ACMILBC::~ACMILBC() {
return;
}
WebRtc_Word16
ACMILBC::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMILBC::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
return -1;
}
WebRtc_Word16
ACMILBC::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMILBC::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return -1;
}
WebRtc_Word16
ACMILBC::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMILBC::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word16
ACMILBC::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMILBC::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word32
ACMILBC::CodecDef(
WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */)
{
WebRtc_Word32 ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */) {
return -1;
}
ACMGenericCodec*
ACMILBC::CreateInstance(void)
{
ACMGenericCodec* ACMILBC::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMILBC::InternalCreateEncoder()
{
WebRtc_Word16 ACMILBC::InternalCreateEncoder() {
return -1;
}
void
ACMILBC::DestructEncoderSafe()
{
void ACMILBC::DestructEncoderSafe() {
return;
}
WebRtc_Word16
ACMILBC::InternalCreateDecoder()
{
WebRtc_Word16 ACMILBC::InternalCreateDecoder() {
return -1;
}
void
ACMILBC::DestructDecoderSafe()
{
void ACMILBC::DestructDecoderSafe() {
return;
}
void
ACMILBC::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMILBC::InternalDestructEncoderInst(void* /* ptrInst */) {
return;
}
WebRtc_Word16
ACMILBC::SetBitRateSafe(const WebRtc_Word32 /* rate */)
{
WebRtc_Word16 ACMILBC::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
return -1;
}
#else //===================== Actual Implementation =======================
ACMILBC::ACMILBC(
WebRtc_Word16 codecID):
_encoderInstPtr(NULL),
_decoderInstPtr(NULL)
{
ACMILBC::ACMILBC(WebRtc_Word16 codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL) {
_codecID = codecID;
return;
}
ACMILBC::~ACMILBC()
{
if(_encoderInstPtr != NULL)
{
ACMILBC::~ACMILBC() {
if (_encoderInstPtr != NULL) {
WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
_encoderInstPtr = NULL;
}
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
_decoderInstPtr = NULL;
}
return;
}
WebRtc_Word16
ACMILBC::InternalEncode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte)
{
WebRtc_Word16 ACMILBC::InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) {
*bitStreamLenByte = WebRtcIlbcfix_Encode(_encoderInstPtr,
&_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
if (*bitStreamLenByte < 0)
{
&_inAudio[_inAudioIxRead],
_frameLenSmpl,
(WebRtc_Word16*) bitStream);
if (*bitStreamLenByte < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalEncode: error in encode for ILBC");
return -1;
@ -179,78 +127,51 @@ ACMILBC::InternalEncode(
return *bitStreamLenByte;
}
WebRtc_Word16
ACMILBC::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMILBC::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return 0;
}
WebRtc_Word16
ACMILBC::InternalInitEncoder(
WebRtcACMCodecParams* codecParams)
{
WebRtc_Word16 ACMILBC::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
// initialize with a correct processing block length
if((160 == (codecParams->codecInstant).pacsize) ||
(320 == (codecParams->codecInstant).pacsize))
{
if ((160 == (codecParams->codecInstant).pacsize) ||
(320 == (codecParams->codecInstant).pacsize)) {
// processing block of 20ms
return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
}
else if((240 == (codecParams->codecInstant).pacsize) ||
(480 == (codecParams->codecInstant).pacsize))
{
} else if ((240 == (codecParams->codecInstant).pacsize) ||
(480 == (codecParams->codecInstant).pacsize)) {
// processing block of 30ms
return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
}
else
{
} else {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalInitEncoder: invalid processing block");
return -1;
}
}
WebRtc_Word16
ACMILBC::InternalInitDecoder(
WebRtcACMCodecParams* codecParams)
{
WebRtc_Word16 ACMILBC::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
// initialize with a correct processing block length
if((160 == (codecParams->codecInstant).pacsize) ||
(320 == (codecParams->codecInstant).pacsize))
{
if ((160 == (codecParams->codecInstant).pacsize) ||
(320 == (codecParams->codecInstant).pacsize)) {
// processing block of 20ms
return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 20);
}
else if((240 == (codecParams->codecInstant).pacsize) ||
(480 == (codecParams->codecInstant).pacsize))
{
} else if ((240 == (codecParams->codecInstant).pacsize) ||
(480 == (codecParams->codecInstant).pacsize)) {
// processing block of 30ms
return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 30);
}
else
{
} else {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalInitDecoder: invalid processing block");
return -1;
}
}
WebRtc_Word32
ACMILBC::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
WebRtc_Word32 ACMILBC::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) {
if (!_decoderInitialized) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"CodeDef: decoder not initialized for ILBC");
return -1;
@ -259,25 +180,18 @@ ACMILBC::CodecDef(
// "SET_CODEC_PAR" & "SET_ILBC_FUNCTION."
// Then return the structure back to NetEQ to add the codec to it's
// database.
SET_CODEC_PAR((codecDef), kDecoderILBC, codecInst.pltype,
_decoderInstPtr, 8000);
SET_CODEC_PAR((codecDef), kDecoderILBC, codecInst.pltype, _decoderInstPtr,
8000);
SET_ILBC_FUNCTIONS((codecDef));
return 0;
}
ACMGenericCodec*
ACMILBC::CreateInstance(void)
{
ACMGenericCodec* ACMILBC::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMILBC::InternalCreateEncoder()
{
if (WebRtcIlbcfix_EncoderCreate(&_encoderInstPtr) < 0)
{
WebRtc_Word16 ACMILBC::InternalCreateEncoder() {
if (WebRtcIlbcfix_EncoderCreate(&_encoderInstPtr) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalCreateEncoder: cannot create instance for ILBC encoder");
return -1;
@ -285,25 +199,17 @@ ACMILBC::InternalCreateEncoder()
return 0;
}
void
ACMILBC::DestructEncoderSafe()
{
void ACMILBC::DestructEncoderSafe() {
_encoderInitialized = false;
_encoderExist = false;
if(_encoderInstPtr != NULL)
{
if (_encoderInstPtr != NULL) {
WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
_encoderInstPtr = NULL;
}
}
WebRtc_Word16
ACMILBC::InternalCreateDecoder()
{
if (WebRtcIlbcfix_DecoderCreate(&_decoderInstPtr) < 0)
{
WebRtc_Word16 ACMILBC::InternalCreateDecoder() {
if (WebRtcIlbcfix_DecoderCreate(&_decoderInstPtr) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"InternalCreateDecoder: cannot create instance for ILBC decoder");
return -1;
@ -311,45 +217,29 @@ ACMILBC::InternalCreateDecoder()
return 0;
}
void
ACMILBC::DestructDecoderSafe()
{
void ACMILBC::DestructDecoderSafe() {
_decoderInitialized = false;
_decoderExist = false;
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
_decoderInstPtr = NULL;
}
}
void
ACMILBC::InternalDestructEncoderInst(
void* ptrInst)
{
if(ptrInst != NULL)
{
WebRtcIlbcfix_EncoderFree((iLBC_encinst_t_*)ptrInst);
void ACMILBC::InternalDestructEncoderInst(void* ptrInst) {
if (ptrInst != NULL) {
WebRtcIlbcfix_EncoderFree((iLBC_encinst_t_*) ptrInst);
}
return;
}
WebRtc_Word16
ACMILBC::SetBitRateSafe(const WebRtc_Word32 rate)
{
WebRtc_Word16 ACMILBC::SetBitRateSafe(const WebRtc_Word32 rate) {
// Check that rate is valid. No need to store the value
if (rate == 13300)
{
if (rate == 13300) {
WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
}
else if (rate == 15200)
{
} else if (rate == 15200) {
WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
}
else
{
} else {
return -1;
}
_encoderParams.codecInstant.rate = rate;

32
webrtc/modules/audio_coding/main/source/acm_ilbc.h
View File

@ -17,42 +17,33 @@
struct iLBC_encinst_t_;
struct iLBC_decinst_t_;
namespace webrtc
{
namespace webrtc {
class ACMILBC : public ACMGenericCodec
{
public:
class ACMILBC : public ACMGenericCodec {
public:
ACMILBC(WebRtc_Word16 codecID);
~ACMILBC();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
WebRtc_Word16 SetBitRateSafe(
const WebRtc_Word32 rate);
WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
void DestructEncoderSafe();
@ -62,8 +53,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
iLBC_encinst_t_* _encoderInstPtr;
iLBC_decinst_t_* _decoderInstPtr;

793
webrtc/modules/audio_coding/main/source/acm_isac.cc
View File

File diff suppressed because it is too large Load Diff

91
webrtc/modules/audio_coding/main/source/acm_isac.h
View File

@ -13,129 +13,104 @@
#include "acm_generic_codec.h"
namespace webrtc
{
namespace webrtc {
struct ACMISACInst;
enum iSACCodingMode {ADAPTIVE, CHANNEL_INDEPENDENT};
enum iSACCodingMode {
ADAPTIVE,
CHANNEL_INDEPENDENT
};
class ACMISAC : public ACMGenericCodec
{
public:
class ACMISAC : public ACMGenericCodec {
public:
ACMISAC(WebRtc_Word16 codecID);
~ACMISAC();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 DeliverCachedIsacData(
WebRtc_UWord8* bitStream,
WebRtc_Word16 DeliverCachedIsacData(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte,
WebRtc_UWord32* timestamp,
WebRtcACMEncodingType* encodingType,
const WebRtc_UWord16 isacRate,
const WebRtc_UWord8 isacBWestimate);
WebRtc_Word16 DeliverCachedData(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 DeliverCachedData(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */,
WebRtc_UWord32* /* timestamp */,
WebRtcACMEncodingType* /* encodingType */)
{
WebRtcACMEncodingType* /* encodingType */) {
return -1;
}
WebRtc_Word16 UpdateDecoderSampFreq(
WebRtc_Word16 codecId);
WebRtc_Word16 UpdateDecoderSampFreq(WebRtc_Word16 codecId);
WebRtc_Word16 UpdateEncoderSampFreq(
WebRtc_UWord16 sampFreqHz);
WebRtc_Word16 UpdateEncoderSampFreq(WebRtc_UWord16 sampFreqHz);
WebRtc_Word16 EncoderSampFreq(
WebRtc_UWord16& sampFreqHz);
WebRtc_Word16 EncoderSampFreq(WebRtc_UWord16& sampFreqHz);
WebRtc_Word32 ConfigISACBandwidthEstimator(
const WebRtc_UWord8 initFrameSizeMsec,
const WebRtc_UWord16 initRateBitPerSec,
const bool enforceFrameSize);
const WebRtc_UWord16 initRateBitPerSec, const bool enforceFrameSize);
WebRtc_Word32 SetISACMaxPayloadSize(
const WebRtc_UWord16 maxPayloadLenBytes);
WebRtc_Word32 SetISACMaxPayloadSize(const WebRtc_UWord16 maxPayloadLenBytes);
WebRtc_Word32 SetISACMaxRate(
const WebRtc_UWord32 maxRateBitPerSec);
WebRtc_Word32 SetISACMaxRate(const WebRtc_UWord32 maxRateBitPerSec);
WebRtc_Word16 REDPayloadISAC(
const WebRtc_Word32 isacRate,
WebRtc_Word16 REDPayloadISAC(const WebRtc_Word32 isacRate,
const WebRtc_Word16 isacBwEstimate,
WebRtc_UWord8* payload,
WebRtc_Word16* payloadLenBytes);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte, WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
void DestructDecoderSafe();
WebRtc_Word16 SetBitRateSafe(
const WebRtc_Word32 bitRate);
WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 bitRate);
WebRtc_Word32 GetEstimatedBandwidthSafe();
WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
WebRtc_Word32 GetRedPayloadSafe(
WebRtc_UWord8* redPayload,
WebRtc_Word32 GetRedPayloadSafe(WebRtc_UWord8* redPayload,
WebRtc_Word16* payloadBytes);
WebRtc_Word16 InternalCreateEncoder();
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 Transcode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte,
WebRtc_Word16 qBWE,
WebRtc_Word32 rate,
bool isRED);
WebRtc_Word16 Transcode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte, WebRtc_Word16 qBWE,
WebRtc_Word32 rate, bool isRED);
void CurrentRate(WebRtc_Word32& rateBitPerSec);
void UpdateFrameLen();
bool DecoderParamsSafe(
WebRtcACMCodecParams *decParams,
bool DecoderParamsSafe(WebRtcACMCodecParams *decParams,
const WebRtc_UWord8 payloadType);
void SaveDecoderParamSafe(
const WebRtcACMCodecParams* codecParams);
void SaveDecoderParamSafe(const WebRtcACMCodecParams* codecParams);
ACMISACInst* _codecInstPtr;
bool _isEncInitialized;
iSACCodingMode _isacCodingMode;
bool _enforceFrameSize;
@ -144,6 +119,6 @@ protected:
WebRtcACMCodecParams _decoderParams32kHz;
};
} //namespace
} // namespace
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_

95
webrtc/modules/audio_coding/main/source/acm_isac_macros.h
View File

@ -13,59 +13,58 @@
#include "engine_configurations.h"
namespace webrtc
{
namespace webrtc {
#ifdef WEBRTC_CODEC_ISAC
# define ACM_ISAC_CREATE WebRtcIsac_Create
# define ACM_ISAC_FREE WebRtcIsac_Free
# define ACM_ISAC_ENCODERINIT WebRtcIsac_EncoderInit
# define ACM_ISAC_ENCODE WebRtcIsac_Encode
# define ACM_ISAC_DECODERINIT WebRtcIsac_DecoderInit
# define ACM_ISAC_DECODE_BWE WebRtcIsac_UpdateBwEstimate
# define ACM_ISAC_DECODE_B WebRtcIsac_Decode
# define ACM_ISAC_DECODEPLC WebRtcIsac_DecodePlc
# define ACM_ISAC_CONTROL WebRtcIsac_Control
# define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe
# define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen
# define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode
# define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw
# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize
# define ACM_ISAC_SETMAXRATE WebRtcIsac_SetMaxRate
# define ACM_ISAC_GETNEWBITSTREAM WebRtcIsac_GetNewBitStream
# define ACM_ISAC_GETSENDBWE WebRtcIsac_GetDownLinkBwIndex
# define ACM_ISAC_SETBWE WebRtcIsac_UpdateUplinkBw
# define ACM_ISAC_GETBWE WebRtcIsac_ReadBwIndex
# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsac_GetNewFrameLen
# define ACM_ISAC_STRUCT ISACStruct
# define ACM_ISAC_GETENCSAMPRATE WebRtcIsac_EncSampRate
# define ACM_ISAC_GETDECSAMPRATE WebRtcIsac_DecSampRate
#define ACM_ISAC_CREATE WebRtcIsac_Create
#define ACM_ISAC_FREE WebRtcIsac_Free
#define ACM_ISAC_ENCODERINIT WebRtcIsac_EncoderInit
#define ACM_ISAC_ENCODE WebRtcIsac_Encode
#define ACM_ISAC_DECODERINIT WebRtcIsac_DecoderInit
#define ACM_ISAC_DECODE_BWE WebRtcIsac_UpdateBwEstimate
#define ACM_ISAC_DECODE_B WebRtcIsac_Decode
#define ACM_ISAC_DECODEPLC WebRtcIsac_DecodePlc
#define ACM_ISAC_CONTROL WebRtcIsac_Control
#define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe
#define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen
#define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode
#define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw
#define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize
#define ACM_ISAC_SETMAXRATE WebRtcIsac_SetMaxRate
#define ACM_ISAC_GETNEWBITSTREAM WebRtcIsac_GetNewBitStream
#define ACM_ISAC_GETSENDBWE WebRtcIsac_GetDownLinkBwIndex
#define ACM_ISAC_SETBWE WebRtcIsac_UpdateUplinkBw
#define ACM_ISAC_GETBWE WebRtcIsac_ReadBwIndex
#define ACM_ISAC_GETNEWFRAMELEN WebRtcIsac_GetNewFrameLen
#define ACM_ISAC_STRUCT ISACStruct
#define ACM_ISAC_GETENCSAMPRATE WebRtcIsac_EncSampRate
#define ACM_ISAC_GETDECSAMPRATE WebRtcIsac_DecSampRate
#endif
#ifdef WEBRTC_CODEC_ISACFX
# define ACM_ISAC_CREATE WebRtcIsacfix_Create
# define ACM_ISAC_FREE WebRtcIsacfix_Free
# define ACM_ISAC_ENCODERINIT WebRtcIsacfix_EncoderInit
# define ACM_ISAC_ENCODE WebRtcIsacfix_Encode
# define ACM_ISAC_DECODERINIT WebRtcIsacfix_DecoderInit
# define ACM_ISAC_DECODE_BWE WebRtcIsacfix_UpdateBwEstimate
# define ACM_ISAC_DECODE_B WebRtcIsacfix_Decode
# define ACM_ISAC_DECODEPLC WebRtcIsacfix_DecodePlc
# define ACM_ISAC_CONTROL ACMISACFixControl // local Impl
# define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl
# define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen
# define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode
# define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl
# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize
# define ACM_ISAC_SETMAXRATE WebRtcIsacfix_SetMaxRate
# define ACM_ISAC_GETNEWBITSTREAM ACMISACFixGetNewBitstream // local Impl
# define ACM_ISAC_GETSENDBWE ACMISACFixGetSendBWE // local Impl
# define ACM_ISAC_SETBWE WebRtcIsacfix_UpdateUplinkBw
# define ACM_ISAC_GETBWE WebRtcIsacfix_ReadBwIndex
# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsacfix_GetNewFrameLen
# define ACM_ISAC_STRUCT ISACFIX_MainStruct
# define ACM_ISAC_GETENCSAMPRATE ACMISACFixGetEncSampRate // local Impl
# define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl
#define ACM_ISAC_CREATE WebRtcIsacfix_Create
#define ACM_ISAC_FREE WebRtcIsacfix_Free
#define ACM_ISAC_ENCODERINIT WebRtcIsacfix_EncoderInit
#define ACM_ISAC_ENCODE WebRtcIsacfix_Encode
#define ACM_ISAC_DECODERINIT WebRtcIsacfix_DecoderInit
#define ACM_ISAC_DECODE_BWE WebRtcIsacfix_UpdateBwEstimate
#define ACM_ISAC_DECODE_B WebRtcIsacfix_Decode
#define ACM_ISAC_DECODEPLC WebRtcIsacfix_DecodePlc
#define ACM_ISAC_CONTROL ACMISACFixControl // local Impl
#define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl
#define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen
#define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode
#define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl
#define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize
#define ACM_ISAC_SETMAXRATE WebRtcIsacfix_SetMaxRate
#define ACM_ISAC_GETNEWBITSTREAM ACMISACFixGetNewBitstream // local Impl
#define ACM_ISAC_GETSENDBWE ACMISACFixGetSendBWE // local Impl
#define ACM_ISAC_SETBWE WebRtcIsacfix_UpdateUplinkBw
#define ACM_ISAC_GETBWE WebRtcIsacfix_ReadBwIndex
#define ACM_ISAC_GETNEWFRAMELEN WebRtcIsacfix_GetNewFrameLen
#define ACM_ISAC_STRUCT ISACFIX_MainStruct
#define ACM_ISAC_GETENCSAMPRATE ACMISACFixGetEncSampRate // local Impl
#define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl
#endif
} //namespace

746
webrtc/modules/audio_coding/main/source/acm_neteq.cc
View File

File diff suppressed because it is too large Load Diff

80
webrtc/modules/audio_coding/main/source/acm_neteq.h
View File

@ -29,9 +29,8 @@ enum ACMSpeechType;
#define MAX_NUM_SLAVE_NETEQ 1
class ACMNetEQ
{
public:
class ACMNetEQ {
public:
// Constructor of the class
ACMNetEQ();
@ -62,8 +61,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 RecIn(
const WebRtc_UWord8* incomingPayload,
WebRtc_Word32 RecIn(const WebRtc_UWord8* incomingPayload,
const WebRtc_Word32 payloadLength,
const WebRtcRTPHeader& rtpInfo);
@ -78,8 +76,7 @@ public:
// Return value : 0 if ok.
// -1 if NetEQ returned an error.
//
WebRtc_Word32 RecOut(
AudioFrame& audioFrame);
WebRtc_Word32 RecOut(AudioFrame& audioFrame);
//
// AddCodec()
@ -94,9 +91,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 AddCodec(
WebRtcNetEQ_CodecDef *codecDef,
bool toMaster = true);
WebRtc_Word32 AddCodec(WebRtcNetEQ_CodecDef *codecDef, bool toMaster = true);
//
// AllocatePacketBuffer()
@ -109,8 +104,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 AllocatePacketBuffer(
const WebRtcNetEQDecoder* usedCodecs,
WebRtc_Word32 AllocatePacketBuffer(const WebRtcNetEQDecoder* usedCodecs,
WebRtc_Word16 noOfCodecs);
//
@ -123,8 +117,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 SetExtraDelay(
const WebRtc_Word32 delayInMS);
WebRtc_Word32 SetExtraDelay(const WebRtc_Word32 delayInMS);
//
// SetAVTPlayout()
@ -136,8 +129,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 SetAVTPlayout(
const bool enable);
WebRtc_Word32 SetAVTPlayout(const bool enable);
//
// AVTPlayout()
@ -167,8 +159,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 SetPlayoutMode(
const AudioPlayoutMode mode);
WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode);
//
// PlayoutMode()
@ -188,8 +179,7 @@ public:
// Return value : 0 if ok.
// <0 if NetEQ returned an error.
//
WebRtc_Word32 NetworkStatistics(
ACMNetworkStatistics* statistics) const;
WebRtc_Word32 NetworkStatistics(ACMNetworkStatistics* statistics) const;
//
// VADMode()
@ -209,8 +199,7 @@ public:
// Return value : 0 if ok.
// -1 if an error occurred.
//
WebRtc_Word16 SetVADMode(
const ACMVADMode mode);
WebRtc_Word16 SetVADMode(const ACMVADMode mode);
//
// DecodeLock()
@ -218,8 +207,7 @@ public:
//
// Return value : Pointer to the decode lock.
//
RWLockWrapper* DecodeLock() const
{
RWLockWrapper* DecodeLock() const {
return _decodeLock;
}
@ -242,10 +230,7 @@ public:
// Return value : 0 if ok.
// -1 if an error occurred.
//
WebRtc_Word16 RemoveCodec(
WebRtcNetEQDecoder codecIdx,
bool isStereo = false);
WebRtc_Word16 RemoveCodec(WebRtcNetEQDecoder codecIdx, bool isStereo = false);
//
// SetBackgroundNoiseMode()
@ -258,8 +243,7 @@ public:
// Return value : 0 if succeeded,
// -1 if failed to set the mode.
//
WebRtc_Word16 SetBackgroundNoiseMode(
const ACMBackgroundNoiseMode mode);
WebRtc_Word16 SetBackgroundNoiseMode(const ACMBackgroundNoiseMode mode);
//
// BackgroundNoiseMode()
@ -267,30 +251,28 @@ public:
//
// Return value : The mode of background noise.
//
WebRtc_Word16 BackgroundNoiseMode(
ACMBackgroundNoiseMode& mode);
WebRtc_Word16 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode);
void SetUniqueId(
WebRtc_Word32 id);
void SetUniqueId(WebRtc_Word32 id);
WebRtc_Word32 PlayoutTimestamp(
WebRtc_UWord32& timestamp);
WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp);
void SetReceivedStereo(
bool receivedStereo);
void SetReceivedStereo(bool receivedStereo);
WebRtc_UWord8 NumSlaves();
enum JB {masterJB = 0, slaveJB = 1};
enum JB {
masterJB = 0,
slaveJB = 1
};
// Delete all slaves.
void RemoveSlaves();
WebRtc_Word16 AddSlave(
const WebRtcNetEQDecoder* usedCodecs,
WebRtc_Word16 AddSlave(const WebRtcNetEQDecoder* usedCodecs,
WebRtc_Word16 noOfCodecs);
private:
private:
//
// RTPPack()
// Creates a Word16 RTP packet out of the payload data in Word16 and
@ -304,18 +286,13 @@ private:
// Output:
// - rtpPacket : The RTP packet.
//
static void RTPPack(
WebRtc_Word16* rtpPacket,
const WebRtc_Word8* payload,
static void RTPPack(WebRtc_Word16* rtpPacket, const WebRtc_Word8* payload,
const WebRtc_Word32 payloadLengthW8,
const WebRtcRTPHeader& rtpInfo);
void LogError(
const char* neteqFuncName,
const WebRtc_Word16 idx) const;
void LogError(const char* neteqFuncName, const WebRtc_Word16 idx) const;
WebRtc_Word16 InitByIdxSafe(
const WebRtc_Word16 idx);
WebRtc_Word16 InitByIdxSafe(const WebRtc_Word16 idx);
// EnableVAD()
// Enable VAD.
@ -325,8 +302,7 @@ private:
//
WebRtc_Word16 EnableVAD();
WebRtc_Word16 EnableVADByIdxSafe(
const WebRtc_Word16 idx);
WebRtc_Word16 EnableVADByIdxSafe(const WebRtc_Word16 idx);
WebRtc_Word16 AllocatePacketBufferByIdxSafe(
const WebRtcNetEQDecoder* usedCodecs,

3
webrtc/modules/audio_coding/main/source/acm_pcm16b.cc
View File

@ -18,7 +18,7 @@
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_PCM16
#include "pcm16b.h"
#include "pcm16b.h"
#endif
namespace webrtc {
@ -90,7 +90,6 @@ void ACMPCM16B::SplitStereoPacket(uint8_t* /*payload*/,
}
#else //===================== Actual Implementation =======================
ACMPCM16B::ACMPCM16B(WebRtc_Word16 codecID) {
_codecID = codecID;
_samplingFreqHz = ACMCodecDB::CodecFreq(_codecID);

28
webrtc/modules/audio_coding/main/source/acm_pcm16b.h
View File

@ -13,37 +13,30 @@
#include "acm_generic_codec.h"
namespace webrtc
{
namespace webrtc {
class ACMPCM16B : public ACMGenericCodec
{
public:
class ACMPCM16B : public ACMGenericCodec {
public:
ACMPCM16B(WebRtc_Word16 codecID);
~ACMPCM16B();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -54,8 +47,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);

2
webrtc/modules/audio_coding/main/source/acm_pcma.cc
View File

@ -120,7 +120,7 @@ void ACMPCMA::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
// end of the bytestream vector. After looping the data is reordered to:
// l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N),
// where N is the total number of samples.
for (int i = 0; i < *payload_length / 2; i ++) {
for (int i = 0; i < *payload_length / 2; i++) {
right_byte = payload[i + 1];
memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2);
payload[*payload_length - 1] = right_byte;

28
webrtc/modules/audio_coding/main/source/acm_pcma.h
View File

@ -13,37 +13,30 @@
#include "acm_generic_codec.h"
namespace webrtc
{
namespace webrtc {
class ACMPCMA : public ACMGenericCodec
{
public:
class ACMPCMA : public ACMGenericCodec {
public:
ACMPCMA(WebRtc_Word16 codecID);
~ACMPCMA();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -54,8 +47,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
};

2
webrtc/modules/audio_coding/main/source/acm_pcmu.cc
View File

@ -122,7 +122,7 @@ void ACMPCMU::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
// end of the bytestream vector. After looping the data is reordered to:
// l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N),
// where N is the total number of samples.
for (int i = 0; i < *payload_length / 2; i ++) {
for (int i = 0; i < *payload_length / 2; i++) {
right_byte = payload[i + 1];
memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2);
payload[*payload_length - 1] = right_byte;

28
webrtc/modules/audio_coding/main/source/acm_pcmu.h
View File

@ -13,37 +13,30 @@
#include "acm_generic_codec.h"
namespace webrtc
{
namespace webrtc {
class ACMPCMU : public ACMGenericCodec
{
public:
class ACMPCMU : public ACMGenericCodec {
public:
ACMPCMU(WebRtc_Word16 codecID);
~ACMPCMU();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -54,8 +47,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
};

80
webrtc/modules/audio_coding/main/source/acm_red.cc
View File

@ -15,71 +15,48 @@
#include "webrtc_neteq.h"
#include "webrtc_neteq_help_macros.h"
namespace webrtc
{
namespace webrtc {
ACMRED::ACMRED(WebRtc_Word16 codecID)
{
ACMRED::ACMRED(WebRtc_Word16 codecID) {
_codecID = codecID;
}
ACMRED::~ACMRED()
{
ACMRED::~ACMRED() {
return;
}
WebRtc_Word16
ACMRED::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMRED::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
// RED is never used as an encoder
// RED has no instance
return 0;
}
WebRtc_Word16
ACMRED::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMRED::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return 0;
}
WebRtc_Word16
ACMRED::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMRED::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
// This codec does not need initialization,
// RED has no instance
return 0;
}
WebRtc_Word16
ACMRED::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMRED::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
// This codec does not need initialization,
// RED has no instance
return 0;
}
WebRtc_Word32
ACMRED::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
WebRtc_Word32 ACMRED::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) {
if (!_decoderInitialized) {
// Todo:
// log error
return -1;
@ -94,48 +71,31 @@ ACMRED::CodecDef(
return 0;
}
ACMGenericCodec*
ACMRED::CreateInstance(void)
{
ACMGenericCodec* ACMRED::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMRED::InternalCreateEncoder()
{
WebRtc_Word16 ACMRED::InternalCreateEncoder() {
// RED has no instance
return 0;
}
WebRtc_Word16
ACMRED::InternalCreateDecoder()
{
WebRtc_Word16 ACMRED::InternalCreateDecoder() {
// RED has no instance
return 0;
}
void
ACMRED::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMRED::InternalDestructEncoderInst(void* /* ptrInst */) {
// RED has no instance
return;
}
void
ACMRED::DestructEncoderSafe()
{
void ACMRED::DestructEncoderSafe() {
// RED has no instance
return;
}
void ACMRED::DestructDecoderSafe()
{
void ACMRED::DestructDecoderSafe() {
// RED has no instance
return;
}

28
webrtc/modules/audio_coding/main/source/acm_red.h
View File

@ -13,37 +13,30 @@
#include "acm_generic_codec.h"
namespace webrtc
{
namespace webrtc {
class ACMRED : public ACMGenericCodec
{
public:
class ACMRED : public ACMGenericCodec {
public:
ACMRED(WebRtc_Word16 codecID);
~ACMRED();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -54,8 +47,7 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
};
} // namespace webrtc

366
webrtc/modules/audio_coding/main/source/acm_speex.cc
View File

@ -17,35 +17,9 @@
#include "webrtc_neteq_help_macros.h"
#ifdef WEBRTC_CODEC_SPEEX
// NOTE! Speex is not included in the open-source package. The following
// interface file is needed:
//
// /modules/audio_coding/codecs/speex/main/interface/speex_interface.h
//
// The API in the header file should match the one below.
//
// int16_t WebRtcSpeex_CreateEnc(SPEEX_encinst_t **SPEEXenc_inst,
// int32_t fs);
// int16_t WebRtcSpeex_FreeEnc(SPEEX_encinst_t *SPEEXenc_inst);
// int16_t WebRtcSpeex_CreateDec(SPEEX_decinst_t **SPEEXdec_inst,
// int32_t fs,
// int16_t enh_enabled);
// int16_t WebRtcSpeex_FreeDec(SPEEX_decinst_t *SPEEXdec_inst);
// int16_t WebRtcSpeex_Encode(SPEEX_encinst_t *SPEEXenc_inst,
// int16_t *speechIn,
// int32_t rate);
// int16_t WebRtcSpeex_EncoderInit(SPEEX_encinst_t *SPEEXenc_inst,
// int16_t vbr, int16_t complexity,
// int16_t vad_enable);
// int16_t WebRtcSpeex_GetBitstream(SPEEX_encinst_t *SPEEXenc_inst,
// int16_t *encoded);
// int16_t WebRtcSpeex_DecodePlc(SPEEX_decinst_t *SPEEXdec_inst,
// int16_t *decoded, int16_t noOfLostFrames);
// int16_t WebRtcSpeex_Decode(SPEEX_decinst_t *SPEEXdec_inst,
// int16_t *encoded, int16_t len,
// int16_t *decoded, int16_t *speechType);
// int16_t WebRtcSpeex_DecoderInit(SPEEX_decinst_t *SPEEXdec_inst);
#include "speex_interface.h"
// NOTE! Speex is not included in the open-source package. Modify this file or
// your codec API to match the function calls and names of used Speex API file.
#include "speex_interface.h"
#endif
namespace webrtc {
@ -62,153 +36,105 @@ ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID */)
return;
}
ACMSPEEX::~ACMSPEEX()
{
ACMSPEEX::~ACMSPEEX() {
return;
}
WebRtc_Word16
ACMSPEEX::InternalEncode(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */)
{
WebRtc_Word16 ACMSPEEX::InternalEncode(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16* /* bitStreamLenByte */) {
return -1;
}
WebRtc_Word16
ACMSPEEX::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMSPEEX::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return -1;
}
WebRtc_Word16
ACMSPEEX::EnableDTX()
{
WebRtc_Word16 ACMSPEEX::EnableDTX() {
return -1;
}
WebRtc_Word16
ACMSPEEX::DisableDTX()
{
WebRtc_Word16 ACMSPEEX::DisableDTX() {
return -1;
}
WebRtc_Word16
ACMSPEEX::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMSPEEX::InternalInitEncoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word16
ACMSPEEX::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMSPEEX::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
return -1;
}
WebRtc_Word32
ACMSPEEX::CodecDef(
WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */)
{
WebRtc_Word32 ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
const CodecInst& /* codecInst */) {
return -1;
}
ACMGenericCodec*
ACMSPEEX::CreateInstance(void)
{
ACMGenericCodec* ACMSPEEX::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMSPEEX::InternalCreateEncoder()
{
WebRtc_Word16 ACMSPEEX::InternalCreateEncoder() {
return -1;
}
void
ACMSPEEX::DestructEncoderSafe()
{
void ACMSPEEX::DestructEncoderSafe() {
return;
}
WebRtc_Word16
ACMSPEEX::InternalCreateDecoder()
{
WebRtc_Word16 ACMSPEEX::InternalCreateDecoder() {
return -1;
}
void
ACMSPEEX::DestructDecoderSafe()
{
void ACMSPEEX::DestructDecoderSafe() {
return;
}
WebRtc_Word16
ACMSPEEX::SetBitRateSafe(
const WebRtc_Word32 /* rate */)
{
WebRtc_Word16 ACMSPEEX::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
return -1;
}
void
ACMSPEEX::InternalDestructEncoderInst(
void* /* ptrInst */)
{
void ACMSPEEX::InternalDestructEncoderInst(void* /* ptrInst */) {
return;
}
#ifdef UNUSEDSPEEX
WebRtc_Word16
ACMSPEEX::EnableVBR()
{
WebRtc_Word16 ACMSPEEX::EnableVBR() {
return -1;
}
WebRtc_Word16
ACMSPEEX::DisableVBR()
{
WebRtc_Word16 ACMSPEEX::DisableVBR() {
return -1;
}
WebRtc_Word16
ACMSPEEX::SetComplMode(
WebRtc_Word16 mode)
{
WebRtc_Word16 ACMSPEEX::SetComplMode(WebRtc_Word16 mode) {
return -1;
}
#endif
#else //===================== Actual Implementation =======================
ACMSPEEX::ACMSPEEX(WebRtc_Word16 codecID):
_encoderInstPtr(NULL),
_decoderInstPtr(NULL)
{
ACMSPEEX::ACMSPEEX(WebRtc_Word16 codecID)
: _encoderInstPtr(NULL),
_decoderInstPtr(NULL) {
_codecID = codecID;
// Set sampling frequency, frame size and rate Speex
if(_codecID == ACMCodecDB::kSPEEX8)
{
if (_codecID == ACMCodecDB::kSPEEX8) {
_samplingFrequency = 8000;
_samplesIn20MsAudio = 160;
_encodingRate = 11000;
}
else if(_codecID == ACMCodecDB::kSPEEX16)
{
} else if (_codecID == ACMCodecDB::kSPEEX16) {
_samplingFrequency = 16000;
_samplesIn20MsAudio = 320;
_encodingRate = 22000;
}
else
{
} else {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Wrong codec id for Speex.");
@ -225,32 +151,25 @@ _decoderInstPtr(NULL)
return;
}
ACMSPEEX::~ACMSPEEX()
{
if(_encoderInstPtr != NULL)
{
ACMSPEEX::~ACMSPEEX() {
if (_encoderInstPtr != NULL) {
WebRtcSpeex_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
if(_decoderInstPtr != NULL)
{
if (_decoderInstPtr != NULL) {
WebRtcSpeex_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
}
return;
}
WebRtc_Word16
ACMSPEEX::InternalEncode(
WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte)
{
WebRtc_Word16 ACMSPEEX::InternalEncode(WebRtc_UWord8* bitStream,
WebRtc_Word16* bitStreamLenByte) {
WebRtc_Word16 status;
WebRtc_Word16 numEncodedSamples = 0;
WebRtc_Word16 n = 0;
while( numEncodedSamples < _frameLenSmpl)
{
while (numEncodedSamples < _frameLenSmpl) {
status = WebRtcSpeex_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
_encodingRate);
@ -259,95 +178,75 @@ ACMSPEEX::InternalEncode(
_inAudioIxRead += _samplesIn20MsAudio;
numEncodedSamples += _samplesIn20MsAudio;
if(status < 0)
{
if (status < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Error in Speex encoder");
return status;
}
// Update VAD, if internal DTX is used
if(_hasInternalDTX && _dtxEnabled)
{
if (_hasInternalDTX && _dtxEnabled) {
_vadLabel[n++] = status;
_vadLabel[n++] = status;
}
if(status == 0)
{
if (status == 0) {
// This frame is detected as inactive. We need send whatever
// encoded so far.
*bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
(WebRtc_Word16*)bitStream);
(WebRtc_Word16*) bitStream);
return *bitStreamLenByte;
}
}
*bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
(WebRtc_Word16*)bitStream);
(WebRtc_Word16*) bitStream);
return *bitStreamLenByte;
}
WebRtc_Word16
ACMSPEEX::DecodeSafe(
WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 ACMSPEEX::DecodeSafe(WebRtc_UWord8* /* bitStream */,
WebRtc_Word16 /* bitStreamLenByte */,
WebRtc_Word16* /* audio */,
WebRtc_Word16* /* audioSamples */,
WebRtc_Word8* /* speechType */)
{
WebRtc_Word8* /* speechType */) {
return 0;
}
WebRtc_Word16
ACMSPEEX::EnableDTX()
{
if(_dtxEnabled)
{
WebRtc_Word16 ACMSPEEX::EnableDTX() {
if (_dtxEnabled) {
return 0;
}
else if(_encoderExist) // check if encoder exist
{
} else if (_encoderExist) { // check if encoder exist
// enable DTX
if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 1) < 0)
{
if (WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1 : 0),
_complMode, 1) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot enable DTX for Speex");
return -1;
}
_dtxEnabled = true;
return 0;
}
else
{
} else {
return -1;
}
return 0;
}
WebRtc_Word16
ACMSPEEX::DisableDTX()
{
if(!_dtxEnabled)
{
WebRtc_Word16 ACMSPEEX::DisableDTX() {
if (!_dtxEnabled) {
return 0;
}
else if(_encoderExist) // check if encoder exist
{
} else if (_encoderExist) { // check if encoder exist
// disable DTX
if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 0) < 0)
{
if (WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1 : 0),
_complMode, 0) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot disable DTX for Speex");
return -1;
}
_dtxEnabled = false;
return 0;
}
else
{
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}
@ -355,20 +254,19 @@ ACMSPEEX::DisableDTX()
return 0;
}
WebRtc_Word16
ACMSPEEX::InternalInitEncoder(
WebRtcACMCodecParams* codecParams)
{
WebRtc_Word16 ACMSPEEX::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
// sanity check
if (_encoderInstPtr == NULL)
{
if (_encoderInstPtr == NULL) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot initialize Speex encoder, instance does not exist");
return -1;
}
WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
status += (WebRtcSpeex_EncoderInit(_encoderInstPtr, _vbrEnabled, _complMode, ((codecParams->enableDTX)? 1:0)) < 0)? -1:0;
status +=
(WebRtcSpeex_EncoderInit(_encoderInstPtr, _vbrEnabled, _complMode,
((codecParams->enableDTX) ? 1 : 0)) < 0) ?
-1 : 0;
if (status >= 0) {
return 0;
@ -379,20 +277,17 @@ ACMSPEEX::InternalInitEncoder(
}
}
WebRtc_Word16
ACMSPEEX::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */)
{
WebRtc_Word16 ACMSPEEX::InternalInitDecoder(
WebRtcACMCodecParams* /* codecParams */) {
WebRtc_Word16 status;
// sanity check
if (_decoderInstPtr == NULL)
{
if (_decoderInstPtr == NULL) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot initialize Speex decoder, instance does not exist");
return -1;
}
status = ((WebRtcSpeex_DecoderInit(_decoderInstPtr) < 0)? -1:0);
status = ((WebRtcSpeex_DecoderInit(_decoderInstPtr) < 0) ? -1 : 0);
if (status >= 0) {
return 0;
@ -403,13 +298,9 @@ ACMSPEEX::InternalInitDecoder(
}
}
WebRtc_Word32
ACMSPEEX::CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst)
{
if (!_decoderInitialized)
{
WebRtc_Word32 ACMSPEEX::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst) {
if (!_decoderInitialized) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Error, Speex decoder is not initialized");
return -1;
@ -420,22 +311,18 @@ ACMSPEEX::CodecDef(
// Then call NetEQ to add the codec to its
// database.
switch(_samplingFrequency)
{
case 8000:
{
switch (_samplingFrequency) {
case 8000: {
SET_CODEC_PAR((codecDef), kDecoderSPEEX_8, codecInst.pltype,
_decoderInstPtr, 8000);
break;
}
case 16000:
{
case 16000: {
SET_CODEC_PAR((codecDef), kDecoderSPEEX_16, codecInst.pltype,
_decoderInstPtr, 16000);
break;
}
default:
{
default: {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Unsupported sampling frequency for Speex");
@ -447,23 +334,16 @@ ACMSPEEX::CodecDef(
return 0;
}
ACMGenericCodec*
ACMSPEEX::CreateInstance(void)
{
ACMGenericCodec* ACMSPEEX::CreateInstance(void) {
return NULL;
}
WebRtc_Word16
ACMSPEEX::InternalCreateEncoder()
{
WebRtc_Word16 ACMSPEEX::InternalCreateEncoder() {
return WebRtcSpeex_CreateEnc(&_encoderInstPtr, _samplingFrequency);
}
void
ACMSPEEX::DestructEncoderSafe()
{
if(_encoderInstPtr != NULL)
{
void ACMSPEEX::DestructEncoderSafe() {
if (_encoderInstPtr != NULL) {
WebRtcSpeex_FreeEnc(_encoderInstPtr);
_encoderInstPtr = NULL;
}
@ -473,18 +353,12 @@ ACMSPEEX::DestructEncoderSafe()
_encodingRate = 0;
}
WebRtc_Word16
ACMSPEEX::InternalCreateDecoder()
{
WebRtc_Word16 ACMSPEEX::InternalCreateDecoder() {
return WebRtcSpeex_CreateDec(&_decoderInstPtr, _samplingFrequency, 1);
}
void
ACMSPEEX::DestructDecoderSafe()
{
if(_decoderInstPtr != NULL)
{
void ACMSPEEX::DestructDecoderSafe() {
if (_decoderInstPtr != NULL) {
WebRtcSpeex_FreeDec(_decoderInstPtr);
_decoderInstPtr = NULL;
}
@ -493,10 +367,7 @@ ACMSPEEX::DestructDecoderSafe()
_decoderInitialized = false;
}
WebRtc_Word16
ACMSPEEX::SetBitRateSafe(
const WebRtc_Word32 rate)
{
WebRtc_Word16 ACMSPEEX::SetBitRateSafe(const WebRtc_Word32 rate) {
// Check if changed rate
if (rate == _encodingRate) {
return 0;
@ -513,14 +384,9 @@ ACMSPEEX::SetBitRateSafe(
return 0;
}
void
ACMSPEEX::InternalDestructEncoderInst(
void* ptrInst)
{
if(ptrInst != NULL)
{
WebRtcSpeex_FreeEnc((SPEEX_encinst_t_*)ptrInst);
void ACMSPEEX::InternalDestructEncoderInst(void* ptrInst) {
if (ptrInst != NULL) {
WebRtcSpeex_FreeEnc((SPEEX_encinst_t_*) ptrInst);
}
return;
}
@ -529,18 +395,14 @@ ACMSPEEX::InternalDestructEncoderInst(
// This API is currently not in use. If requested to be able to enable/disable VBR
// an ACM API need to be added.
WebRtc_Word16
ACMSPEEX::EnableVBR()
{
if(_vbrEnabled)
{
WebRtc_Word16 ACMSPEEX::EnableVBR() {
if (_vbrEnabled) {
return 0;
}
else if(_encoderExist) // check if encoder exist
} else if (_encoderExist) // check if encoder exist
{
// enable Variable Bit Rate (VBR)
if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 1, _complMode, (_dtxEnabled? 1:0)) < 0)
{
if (WebRtcSpeex_EncoderInit(_encoderInstPtr, 1, _complMode,
(_dtxEnabled ? 1 : 0)) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot enable VBR mode for Speex");
@ -548,28 +410,20 @@ ACMSPEEX::EnableVBR()
}
_vbrEnabled = true;
return 0;
}
else
{
} else {
return -1;
}
}
// This API is currently not in use. If requested to be able to enable/disable VBR
// an ACM API need to be added.
WebRtc_Word16
ACMSPEEX::DisableVBR()
{
if(!_vbrEnabled)
{
// This API is currently not in use. If requested to be able to enable/disable
// VBR an ACM API need to be added.
WebRtc_Word16 ACMSPEEX::DisableVBR() {
if (!_vbrEnabled) {
return 0;
}
else if(_encoderExist) // check if encoder exist
{
} else if (_encoderExist) { // check if encoder exist
// disable DTX
if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, _complMode, (_dtxEnabled? 1:0)) < 0)
{
if (WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, _complMode,
(_dtxEnabled ? 1 : 0)) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Cannot disable DTX for Speex");
@ -577,9 +431,7 @@ ACMSPEEX::DisableVBR()
}
_vbrEnabled = false;
return 0;
}
else
{
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}
@ -587,29 +439,21 @@ ACMSPEEX::DisableVBR()
// This API is currently not in use. If requested to be able to set complexity
// an ACM API need to be added.
WebRtc_Word16
ACMSPEEX::SetComplMode(
WebRtc_Word16 mode)
{
WebRtc_Word16 ACMSPEEX::SetComplMode(WebRtc_Word16 mode) {
// Check if new mode
if(mode == _complMode)
{
if (mode == _complMode) {
return 0;
}
else if(_encoderExist) // check if encoder exist
{
} else if (_encoderExist) { // check if encoder exist
// Set new mode
if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, mode, (_dtxEnabled? 1:0)) < 0)
{
if (WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, mode, (_dtxEnabled ? 1 : 0))
< 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
"Error in complexity mode for Speex");
return -1;
}
_complMode = mode;
return 0;
}
else
{
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}

31
webrtc/modules/audio_coding/main/source/acm_speex.h
View File

@ -19,34 +19,28 @@ struct SPEEX_decinst_t_;
namespace webrtc {
class ACMSPEEX : public ACMGenericCodec
{
public:
class ACMSPEEX : public ACMGenericCodec {
public:
ACMSPEEX(WebRtc_Word16 codecID);
~ACMSPEEX();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(
WebRtc_UWord8* bitstream,
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(
WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(
WebRtc_UWord8* bitStream,
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(
WebRtcNetEQ_CodecDef& codecDef,
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
@ -57,11 +51,9 @@ protected:
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(
void* ptrInst);
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 SetBitRateSafe(
const WebRtc_Word32 rate);
WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
WebRtc_Word16 EnableDTX();
@ -72,8 +64,7 @@ protected:
WebRtc_Word16 DisableVBR();
WebRtc_Word16 SetComplMode(
WebRtc_Word16 mode);
WebRtc_Word16 SetComplMode(WebRtc_Word16 mode);
#endif
SPEEX_encinst_t_* _encoderInstPtr;

44
webrtc/modules/audio_coding/main/source/audio_coding_module.cc
View File

@ -8,51 +8,38 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "acm_dtmf_detection.h"
#include "audio_coding_module.h"
#include "audio_coding_module_impl.h"
#include "trace.h"
namespace webrtc
{
namespace webrtc {
// Create module
AudioCodingModule*
AudioCodingModule::Create(
const WebRtc_Word32 id)
{
AudioCodingModule* AudioCodingModule::Create(const WebRtc_Word32 id) {
return new AudioCodingModuleImpl(id);
}
// Destroy module
void
AudioCodingModule::Destroy(
AudioCodingModule* module)
{
delete static_cast<AudioCodingModuleImpl*> (module);
void AudioCodingModule::Destroy(AudioCodingModule* module) {
delete static_cast<AudioCodingModuleImpl*>(module);
}
// Get number of supported codecs
WebRtc_UWord8 AudioCodingModule::NumberOfCodecs()
{
WebRtc_UWord8 AudioCodingModule::NumberOfCodecs() {
return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs);
}
// Get supported codec param with id
WebRtc_Word32
AudioCodingModule::Codec(
const WebRtc_UWord8 listId,
CodecInst& codec)
{
WebRtc_Word32 AudioCodingModule::Codec(const WebRtc_UWord8 listId,
CodecInst& codec) {
// Get the codec settings for the codec with the given list ID
return ACMCodecDB::Codec(listId, &codec);
}
// Get supported codec Param with name, frequency and number of channels.
WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
CodecInst& codec,
int sampling_freq_hz,
CodecInst& codec, int sampling_freq_hz,
int channels) {
int codec_id;
@ -77,28 +64,21 @@ WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
// Get supported codec Index with name, frequency and number of channels.
WebRtc_Word32 AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz,
int channels) {
int sampling_freq_hz, int channels) {
return ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels);
}
// Checks the validity of the parameters of the given codec
bool
AudioCodingModule::IsCodecValid(
const CodecInst& codec)
{
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
int mirrorID;
char errMsg[500];
int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500);
if(codecNumber < 0)
{
if (codecNumber < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, errMsg);
return false;
}
else
{
} else {
return true;
}
}