f7fa6276e2
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
344 lines
9.0 KiB
C++
344 lines
9.0 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "acm_g7291.h"
|
|
#include "acm_common_defs.h"
|
|
#include "acm_neteq.h"
|
|
#include "trace.h"
|
|
#include "webrtc_neteq.h"
|
|
#include "webrtc_neteq_help_macros.h"
|
|
|
|
#ifdef WEBRTC_CODEC_G729_1
|
|
// NOTE! G.729.1 is not included in the open-source package. 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
|
|
|
|
namespace webrtc {
|
|
|
|
#ifndef WEBRTC_CODEC_G729_1
|
|
|
|
ACMG729_1::ACMG729_1(WebRtc_Word16 /* codecID */)
|
|
: _encoderInstPtr(NULL),
|
|
_decoderInstPtr(NULL),
|
|
_myRate(32000),
|
|
_flag8kHz(0),
|
|
_flagG729mode(0) {
|
|
return;
|
|
}
|
|
|
|
ACMG729_1::~ACMG729_1() {
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalEncode(WebRtc_UWord8* /* bitStream */,
|
|
WebRtc_Word16* /* bitStreamLenByte */) {
|
|
return -1;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
|
|
WebRtc_Word16 /* bitStreamLenByte */,
|
|
WebRtc_Word16* /* audio */,
|
|
WebRtc_Word16* /* audioSamples */,
|
|
WebRtc_Word8* /* speechType */) {
|
|
return -1;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalInitEncoder(
|
|
WebRtcACMCodecParams* /* codecParams */) {
|
|
return -1;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalInitDecoder(
|
|
WebRtcACMCodecParams* /* codecParams */) {
|
|
return -1;
|
|
}
|
|
|
|
WebRtc_Word32 ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
|
|
const CodecInst& /* codecInst */) {
|
|
return -1;
|
|
}
|
|
|
|
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
|
|
return NULL;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalCreateEncoder() {
|
|
return -1;
|
|
}
|
|
|
|
void ACMG729_1::DestructEncoderSafe() {
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalCreateDecoder() {
|
|
return -1;
|
|
}
|
|
|
|
void ACMG729_1::DestructDecoderSafe() {
|
|
return;
|
|
}
|
|
|
|
void ACMG729_1::InternalDestructEncoderInst(void* /* ptrInst */) {
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::SetBitRateSafe(const WebRtc_Word32 /*rate*/) {
|
|
return -1;
|
|
}
|
|
|
|
#else //===================== Actual Implementation =======================
|
|
|
|
struct G729_1_inst_t_;
|
|
|
|
ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
|
|
: _encoderInstPtr(NULL),
|
|
_decoderInstPtr(NULL),
|
|
_myRate(32000), // Default rate.
|
|
_flag8kHz(0),
|
|
_flagG729mode(0) {
|
|
// TODO(tlegrand): We should add codecID as a input variable to the
|
|
// constructor of ACMGenericCodec.
|
|
_codecID = codecID;
|
|
return;
|
|
}
|
|
|
|
ACMG729_1::~ACMG729_1() {
|
|
if (_encoderInstPtr != NULL) {
|
|
WebRtcG7291_Free(_encoderInstPtr);
|
|
_encoderInstPtr = NULL;
|
|
}
|
|
if (_decoderInstPtr != NULL) {
|
|
WebRtcG7291_Free(_decoderInstPtr);
|
|
_decoderInstPtr = NULL;
|
|
}
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalEncode(WebRtc_UWord8* bitStream,
|
|
WebRtc_Word16* bitStreamLenByte) {
|
|
|
|
// Initialize before entering the loop
|
|
WebRtc_Word16 noEncodedSamples = 0;
|
|
*bitStreamLenByte = 0;
|
|
|
|
WebRtc_Word16 byteLengthFrame = 0;
|
|
|
|
// Derive number of 20ms frames per encoded packet.
|
|
// [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
|
|
WebRtc_Word16 n20msFrames = (_frameLenSmpl / 320);
|
|
// Byte length for the frame. +1 is for rate information.
|
|
byteLengthFrame = _myRate / (8 * 50) * n20msFrames + (1 - _flagG729mode);
|
|
|
|
// The following might be revised if we have G729.1 Annex C (support for DTX);
|
|
do {
|
|
*bitStreamLenByte = WebRtcG7291_Encode(_encoderInstPtr,
|
|
&_inAudio[_inAudioIxRead],
|
|
(WebRtc_Word16*) bitStream, _myRate,
|
|
n20msFrames);
|
|
|
|
// increment the read index this tell the caller that how far
|
|
// we have gone forward in reading the audio buffer
|
|
_inAudioIxRead += 160;
|
|
|
|
// sanity check
|
|
if (*bitStreamLenByte < 0) {
|
|
// error has happened
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"InternalEncode: Encode error for G729_1");
|
|
*bitStreamLenByte = 0;
|
|
return -1;
|
|
}
|
|
|
|
noEncodedSamples += 160;
|
|
} while (*bitStreamLenByte == 0);
|
|
|
|
// This criteria will change if we have Annex C.
|
|
if (*bitStreamLenByte != byteLengthFrame) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"InternalEncode: Encode error for G729_1");
|
|
*bitStreamLenByte = 0;
|
|
return -1;
|
|
}
|
|
|
|
if (noEncodedSamples != _frameLenSmpl) {
|
|
*bitStreamLenByte = 0;
|
|
return -1;
|
|
}
|
|
|
|
return *bitStreamLenByte;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
|
|
WebRtc_Word16 /* bitStreamLenByte */,
|
|
WebRtc_Word16* /* audio */,
|
|
WebRtc_Word16* /* audioSamples */,
|
|
WebRtc_Word8* /* speechType */) {
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalInitEncoder(
|
|
WebRtcACMCodecParams* codecParams) {
|
|
//set the bit rate and initialize
|
|
_myRate = codecParams->codecInstant.rate;
|
|
return SetBitRateSafe((WebRtc_UWord32) _myRate);
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalInitDecoder(
|
|
WebRtcACMCodecParams* /* codecParams */) {
|
|
if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"InternalInitDecoder: init decoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word32 ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
|
|
const CodecInst& codecInst) {
|
|
if (!_decoderInitialized) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"CodeDef: Decoder uninitialized for G729_1");
|
|
return -1;
|
|
}
|
|
|
|
// Fill up the structure by calling
|
|
// "SET_CODEC_PAR" & "SET_G729_FUNCTION."
|
|
// Then call NetEQ to add the codec to it's
|
|
// database.
|
|
SET_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype, _decoderInstPtr,
|
|
16000);
|
|
SET_G729_1_FUNCTIONS((codecDef));
|
|
return 0;
|
|
}
|
|
|
|
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
|
|
return NULL;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalCreateEncoder() {
|
|
if (WebRtcG7291_Create(&_encoderInstPtr) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"InternalCreateEncoder: create encoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ACMG729_1::DestructEncoderSafe() {
|
|
_encoderExist = false;
|
|
_encoderInitialized = false;
|
|
if (_encoderInstPtr != NULL) {
|
|
WebRtcG7291_Free(_encoderInstPtr);
|
|
_encoderInstPtr = NULL;
|
|
}
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::InternalCreateDecoder() {
|
|
if (WebRtcG7291_Create(&_decoderInstPtr) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"InternalCreateDecoder: create decoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ACMG729_1::DestructDecoderSafe() {
|
|
_decoderExist = false;
|
|
_decoderInitialized = false;
|
|
if (_decoderInstPtr != NULL) {
|
|
WebRtcG7291_Free(_decoderInstPtr);
|
|
_decoderInstPtr = NULL;
|
|
}
|
|
}
|
|
|
|
void ACMG729_1::InternalDestructEncoderInst(void* ptrInst) {
|
|
if (ptrInst != NULL) {
|
|
//WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
|
|
}
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMG729_1::SetBitRateSafe(const WebRtc_Word32 rate) {
|
|
//allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
|
|
// 22000, 24000, 26000, 28000, 30000, 32000};
|
|
// TODO(tlegrand): This check exists in one other place two. Should be
|
|
// possible to reuse code.
|
|
switch (rate) {
|
|
case 8000: {
|
|
_myRate = 8000;
|
|
break;
|
|
}
|
|
case 12000: {
|
|
_myRate = 12000;
|
|
break;
|
|
}
|
|
case 14000: {
|
|
_myRate = 14000;
|
|
break;
|
|
}
|
|
case 16000: {
|
|
_myRate = 16000;
|
|
break;
|
|
}
|
|
case 18000: {
|
|
_myRate = 18000;
|
|
break;
|
|
}
|
|
case 20000: {
|
|
_myRate = 20000;
|
|
break;
|
|
}
|
|
case 22000: {
|
|
_myRate = 22000;
|
|
break;
|
|
}
|
|
case 24000: {
|
|
_myRate = 24000;
|
|
break;
|
|
}
|
|
case 26000: {
|
|
_myRate = 26000;
|
|
break;
|
|
}
|
|
case 28000: {
|
|
_myRate = 28000;
|
|
break;
|
|
}
|
|
case 30000: {
|
|
_myRate = 30000;
|
|
break;
|
|
}
|
|
case 32000: {
|
|
_myRate = 32000;
|
|
break;
|
|
}
|
|
default: {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
|
|
"SetBitRateSafe: Invalid rate G729_1");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
// Re-init with new rate
|
|
if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz,
|
|
_flagG729mode) >= 0) {
|
|
_encoderParams.codecInstant.rate = _myRate;
|
|
return 0;
|
|
} else {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
} // namespace webrtc
|