
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
260 lines
7.1 KiB
C++
260 lines
7.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "acm_gsmfr.h"
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#include "acm_common_defs.h"
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#include "acm_neteq.h"
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#include "trace.h"
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#include "webrtc_neteq.h"
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#include "webrtc_neteq_help_macros.h"
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#ifdef WEBRTC_CODEC_GSMFR
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// NOTE! GSM-FR is not included in the open-source package. Modify this file
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// or your codec API to match the function calls and names of used GSM-FR API
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// file.
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#include "gsmfr_interface.h"
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#endif
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namespace webrtc {
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#ifndef WEBRTC_CODEC_GSMFR
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ACMGSMFR::ACMGSMFR(WebRtc_Word16 /* codecID */)
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: _encoderInstPtr(NULL),
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_decoderInstPtr(NULL) {
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return;
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}
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ACMGSMFR::~ACMGSMFR() {
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return;
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}
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WebRtc_Word16 ACMGSMFR::InternalEncode(WebRtc_UWord8* /* bitStream */,
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WebRtc_Word16* /* bitStreamLenByte */) {
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return -1;
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}
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WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
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WebRtc_Word16 /* bitStreamLenByte */,
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WebRtc_Word16* /* audio */,
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WebRtc_Word16* /* audioSamples */,
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WebRtc_Word8* /* speechType */) {
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return -1;
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}
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WebRtc_Word16 ACMGSMFR::EnableDTX() {
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return -1;
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}
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WebRtc_Word16 ACMGSMFR::DisableDTX() {
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return -1;
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}
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WebRtc_Word16 ACMGSMFR::InternalInitEncoder(
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WebRtcACMCodecParams* /* codecParams */) {
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return -1;
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}
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WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
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WebRtcACMCodecParams* /* codecParams */) {
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return -1;
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}
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WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
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const CodecInst& /* codecInst */) {
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return -1;
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}
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ACMGenericCodec* ACMGSMFR::CreateInstance(void) {
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return NULL;
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}
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WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
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return -1;
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}
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void ACMGSMFR::DestructEncoderSafe() {
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return;
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}
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WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
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return -1;
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}
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void ACMGSMFR::DestructDecoderSafe() {
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return;
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}
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void ACMGSMFR::InternalDestructEncoderInst(void* /* ptrInst */) {
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return;
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}
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#else //===================== Actual Implementation =======================
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ACMGSMFR::ACMGSMFR(WebRtc_Word16 codecID)
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: _encoderInstPtr(NULL),
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_decoderInstPtr(NULL) {
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_codecID = codecID;
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_hasInternalDTX = true;
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return;
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}
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ACMGSMFR::~ACMGSMFR() {
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if (_encoderInstPtr != NULL) {
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WebRtcGSMFR_FreeEnc(_encoderInstPtr);
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_encoderInstPtr = NULL;
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}
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if (_decoderInstPtr != NULL) {
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WebRtcGSMFR_FreeDec(_decoderInstPtr);
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_decoderInstPtr = NULL;
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}
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return;
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}
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WebRtc_Word16 ACMGSMFR::InternalEncode(WebRtc_UWord8* bitStream,
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WebRtc_Word16* bitStreamLenByte) {
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*bitStreamLenByte = WebRtcGSMFR_Encode(_encoderInstPtr,
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&_inAudio[_inAudioIxRead],
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_frameLenSmpl,
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(WebRtc_Word16*) bitStream);
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// increment the read index this tell the caller that how far
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// we have gone forward in reading the audio buffer
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_inAudioIxRead += _frameLenSmpl;
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return *bitStreamLenByte;
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}
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WebRtc_Word16 ACMGSMFR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
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WebRtc_Word16 /* bitStreamLenByte */,
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WebRtc_Word16* /* audio */,
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WebRtc_Word16* /* audioSamples */,
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WebRtc_Word8* /* speechType */) {
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return 0;
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}
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WebRtc_Word16 ACMGSMFR::EnableDTX() {
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if (_dtxEnabled) {
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return 0;
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} else if (_encoderExist) {
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if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, 1) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"EnableDTX: cannot init encoder for GSMFR");
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return -1;
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}
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_dtxEnabled = true;
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return 0;
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} else {
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return -1;
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}
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}
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WebRtc_Word16 ACMGSMFR::DisableDTX() {
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if (!_dtxEnabled) {
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return 0;
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} else if (_encoderExist) {
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if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, 0) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"DisableDTX: cannot init encoder for GSMFR");
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return -1;
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}
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_dtxEnabled = false;
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return 0;
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} else {
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// encoder doesn't exists, therefore disabling is harmless
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return 0;
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}
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}
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WebRtc_Word16 ACMGSMFR::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
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if (WebRtcGSMFR_EncoderInit(_encoderInstPtr,
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((codecParams->enableDTX) ? 1 : 0)) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"InternalInitEncoder: cannot init encoder for GSMFR");
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}
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return 0;
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}
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WebRtc_Word16 ACMGSMFR::InternalInitDecoder(
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WebRtcACMCodecParams* /* codecParams */) {
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if (WebRtcGSMFR_DecoderInit(_decoderInstPtr) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"InternalInitDecoder: cannot init decoder for GSMFR");
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return -1;
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}
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return 0;
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}
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WebRtc_Word32 ACMGSMFR::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
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const CodecInst& codecInst) {
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if (!_decoderInitialized) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"CodecDef: decoder is not initialized for GSMFR");
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return -1;
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}
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// Fill up the structure by calling
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// "SET_CODEC_PAR" & "SET_GSMFR_FUNCTION."
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// Then call NetEQ to add the codec to it's
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// database.
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SET_CODEC_PAR((codecDef), kDecoderGSMFR, codecInst.pltype, _decoderInstPtr,
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8000);
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SET_GSMFR_FUNCTIONS((codecDef));
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return 0;
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}
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ACMGenericCodec* ACMGSMFR::CreateInstance(void) {
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return NULL;
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}
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WebRtc_Word16 ACMGSMFR::InternalCreateEncoder() {
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if (WebRtcGSMFR_CreateEnc(&_encoderInstPtr) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"InternalCreateEncoder: cannot create instance for GSMFR encoder");
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return -1;
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}
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return 0;
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}
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void ACMGSMFR::DestructEncoderSafe() {
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if (_encoderInstPtr != NULL) {
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WebRtcGSMFR_FreeEnc(_encoderInstPtr);
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_encoderInstPtr = NULL;
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}
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_encoderExist = false;
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_encoderInitialized = false;
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}
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WebRtc_Word16 ACMGSMFR::InternalCreateDecoder() {
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if (WebRtcGSMFR_CreateDec(&_decoderInstPtr) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
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"InternalCreateDecoder: cannot create instance for GSMFR decoder");
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return -1;
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}
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return 0;
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}
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void ACMGSMFR::DestructDecoderSafe() {
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if (_decoderInstPtr != NULL) {
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WebRtcGSMFR_FreeDec(_decoderInstPtr);
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_decoderInstPtr = NULL;
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}
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_decoderExist = false;
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_decoderInitialized = false;
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}
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void ACMGSMFR::InternalDestructEncoderInst(void* ptrInst) {
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if (ptrInst != NULL) {
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WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*) ptrInst);
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}
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return;
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}
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#endif
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} // namespace webrtc
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