mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
162 lines
4.5 KiB
C++
162 lines
4.5 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 "webrtc/modules/audio_coding/main/acm2/acm_gsmfr.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 "webrtc/modules/audio_coding/main/codecs/gsmfr/interface/gsmfr_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#endif
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namespace webrtc {
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namespace acm2 {
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#ifndef WEBRTC_CODEC_GSMFR
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ACMGSMFR::ACMGSMFR(int16_t /* codec_id */) : encoder_inst_ptr_(NULL) {}
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ACMGSMFR::~ACMGSMFR() { return; }
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int16_t ACMGSMFR::InternalEncode(uint8_t* /* bitstream */,
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int16_t* /* bitstream_len_byte */) {
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return -1;
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}
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int16_t ACMGSMFR::EnableDTX() { return -1; }
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int16_t ACMGSMFR::DisableDTX() { return -1; }
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int16_t ACMGSMFR::InternalInitEncoder(
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WebRtcACMCodecParams* /* codec_params */) {
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return -1;
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}
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ACMGenericCodec* ACMGSMFR::CreateInstance(void) { return NULL; }
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int16_t ACMGSMFR::InternalCreateEncoder() { return -1; }
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void ACMGSMFR::DestructEncoderSafe() { return; }
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void ACMGSMFR::InternalDestructEncoderInst(void* /* ptr_inst */) {
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return;
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}
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#else //===================== Actual Implementation =======================
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ACMGSMFR::ACMGSMFR(int16_t codec_id)
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: codec_id_(codec_id),
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has_internal_dtx_(true),
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encoder_inst_ptr_(NULL) {}
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ACMGSMFR::~ACMGSMFR() {
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if (encoder_inst_ptr_ != NULL) {
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WebRtcGSMFR_FreeEnc(encoder_inst_ptr_);
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encoder_inst_ptr_ = NULL;
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}
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return;
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}
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int16_t ACMGSMFR::InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) {
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*bitstream_len_byte = WebRtcGSMFR_Encode(
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encoder_inst_ptr_, &in_audio_[in_audio_ix_read_], frame_len_smpl_,
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reinterpret_cast<int16_t*>(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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in_audio_ix_read_ += frame_len_smpl_;
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return *bitstream_len_byte;
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}
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int16_t ACMGSMFR::EnableDTX() {
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if (dtx_enabled_) {
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return 0;
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} else if (encoder_exist_) {
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if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 1) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"EnableDTX: cannot init encoder for GSMFR");
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return -1;
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}
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dtx_enabled_ = 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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int16_t ACMGSMFR::DisableDTX() {
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if (!dtx_enabled_) {
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return 0;
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} else if (encoder_exist_) {
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if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_, 0) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"DisableDTX: cannot init encoder for GSMFR");
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return -1;
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}
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dtx_enabled_ = 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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int16_t ACMGSMFR::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
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if (WebRtcGSMFR_EncoderInit(encoder_inst_ptr_,
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((codec_params->enable_dtx) ? 1 : 0)) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError,
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webrtc::kTraceAudioCoding,
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unique_id_,
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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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ACMGenericCodec* ACMGSMFR::CreateInstance(void) { return NULL; }
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int16_t ACMGSMFR::InternalCreateEncoder() {
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if (WebRtcGSMFR_CreateEnc(&encoder_inst_ptr_) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError,
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webrtc::kTraceAudioCoding,
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unique_id_,
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"InternalCreateEncoder: cannot create instance for GSMFR "
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"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 (encoder_inst_ptr_ != NULL) {
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WebRtcGSMFR_FreeEnc(encoder_inst_ptr_);
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encoder_inst_ptr_ = NULL;
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}
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encoder_exist_ = false;
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encoder_initialized_ = false;
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}
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void ACMGSMFR::InternalDestructEncoderInst(void* ptr_inst) {
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if (ptr_inst != NULL) {
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WebRtcGSMFR_FreeEnc(static_cast<GSMFR_encinst_t_*>(ptr_inst));
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}
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return;
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}
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#endif
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} // namespace acm2
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} // namespace webrtc
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