session-android/jni/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc

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/*
* 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 {
namespace acm2 {
#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 acm2
} // namespace webrtc