Files
platform-external-webrtc/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc
turaj@webrtc.org 48af652ea5 Prepare to compile ACM1 and ACM2.
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 23:06:59 +00:00

158 lines
4.4 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 "webrtc/modules/audio_coding/main/acm2/acm_gsmfr.h"
#ifdef WEBRTC_CODEC_GSMFR
// 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 "webrtc/modules/audio_coding/main/codecs/gsmfr/interface/gsmfr_interface.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
#include "webrtc/system_wrappers/interface/trace.h"
#endif
namespace webrtc {
#ifndef WEBRTC_CODEC_GSMFR
ACMGSMFR::ACMGSMFR(int16_t /* codec_id */) : encoder_inst_ptr_(NULL) {}
ACMGSMFR::~ACMGSMFR() { return; }
int16_t ACMGSMFR::InternalEncode(uint8_t* /* bitstream */,
int16_t* /* bitstream_len_byte */) {
return -1;
}
int16_t ACMGSMFR::EnableDTX() { return -1; }
int16_t ACMGSMFR::DisableDTX() { return -1; }
int16_t ACMGSMFR::InternalInitEncoder(
WebRtcACMCodecParams* /* codec_params */) {
return -1;
}
ACMGenericCodec* ACMGSMFR::CreateInstance(void) { return NULL; }
int16_t ACMGSMFR::InternalCreateEncoder() { return -1; }
void ACMGSMFR::DestructEncoderSafe() { return; }
void ACMGSMFR::InternalDestructEncoderInst(void* /* ptr_inst */) {
return;
}
#else //===================== Actual Implementation =======================
ACMGSMFR::ACMGSMFR(int16_t codec_id)
: codec_id_(codec_id),
has_internal_dtx_(true),
encoder_inst_ptr_(NULL) {}
ACMGSMFR::~ACMGSMFR() {
if (encoder_inst_ptr_ != NULL) {
WebRtcGSMFR_FreeEnc(encoder_inst_ptr_);
encoder_inst_ptr_ = NULL;
}
return;
}
int16_t ACMGSMFR::InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) {
*bitstream_len_byte = WebRtcGSMFR_Encode(
encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], frame_len_smpl_,
reinterpret_cast<int16_t*>(bitstream));
// increment the read index this tell the caller that how far
// we have gone forward in reading the audio buffer
in_audio_ix_read_ += frame_len_smpl_;
return *bitstream_len_byte;
}
int16_t ACMGSMFR::EnableDTX() {
if (dtx_enabled_) {
return 0;
} else if (encoder_exist_) {
if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 1) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
"EnableDTX: cannot init encoder for GSMFR");
return -1;
}
dtx_enabled_ = true;
return 0;
} else {
return -1;
}
}
int16_t ACMGSMFR::DisableDTX() {
if (!dtx_enabled_) {
return 0;
} else if (encoder_exist_) {
if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 0) < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
"DisableDTX: cannot init encoder for GSMFR");
return -1;
}
dtx_enabled_ = false;
return 0;
} else {
// encoder doesn't exists, therefore disabling is harmless
return 0;
}
}
int16_t ACMGSMFR::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_,
((codec_params->enable_dtx) ? 1 : 0)) < 0) {
WEBRTC_TRACE(webrtc::kTraceError,
webrtc::kTraceAudioCoding,
unique_id_,
"InternalInitEncoder: cannot init encoder for GSMFR");
}
return 0;
}
ACMGenericCodec* ACMGSMFR::CreateInstance(void) { return NULL; }
int16_t ACMGSMFR::InternalCreateEncoder() {
if (WebRtcGSMFR_CreateEnc(&encoder_inst_ptr_) < 0) {
WEBRTC_TRACE(webrtc::kTraceError,
webrtc::kTraceAudioCoding,
unique_id_,
"InternalCreateEncoder: cannot create instance for GSMFR "
"encoder");
return -1;
}
return 0;
}
void ACMGSMFR::DestructEncoderSafe() {
if (encoder_inst_ptr_ != NULL) {
WebRtcGSMFR_FreeEnc(encoder_inst_ptr_);
encoder_inst_ptr_ = NULL;
}
encoder_exist_ = false;
encoder_initialized_ = false;
}
void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) {
if (ptr_inst != NULL) {
WebRtcGSMFR_FreeEnc(static_cast<GSMFR_encinst_t_*>(ptr_inst));
}
return;
}
#endif
} // namespace webrtc