mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
337 lines
11 KiB
C++
337 lines
11 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_g7221c.h"
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#ifdef WEBRTC_CODEC_G722_1C
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// NOTE! G.722.1C is not included in the open-source package. The following
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// interface file is needed:
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#include "webrtc/modules/audio_coding/main/codecs/g7221c/interface/g7221c_interface.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.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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// The API in the header file should match the one below.
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//
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// int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_CreateEnc48(G722_1C_48_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_CreateDec24(G722_1C_24_decinst_t_** dec_inst);
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// int16_t WebRtcG7221C_CreateDec32(G722_1C_32_decinst_t_** dec_inst);
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// int16_t WebRtcG7221C_CreateDec48(G722_1C_48_decinst_t_** dec_inst);
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//
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// int16_t WebRtcG7221C_FreeEnc24(G722_1C_24_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_FreeEnc32(G722_1C_32_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_FreeEnc48(G722_1C_48_encinst_t_** enc_inst);
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// int16_t WebRtcG7221C_FreeDec24(G722_1C_24_decinst_t_** dec_inst);
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// int16_t WebRtcG7221C_FreeDec32(G722_1C_32_decinst_t_** dec_inst);
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// int16_t WebRtcG7221C_FreeDec48(G722_1C_48_decinst_t_** dec_inst);
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//
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// int16_t WebRtcG7221C_EncoderInit24(G722_1C_24_encinst_t_* enc_inst);
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// int16_t WebRtcG7221C_EncoderInit32(G722_1C_32_encinst_t_* enc_inst);
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// int16_t WebRtcG7221C_EncoderInit48(G722_1C_48_encinst_t_* enc_inst);
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// int16_t WebRtcG7221C_DecoderInit24(G722_1C_24_decinst_t_* dec_inst);
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// int16_t WebRtcG7221C_DecoderInit32(G722_1C_32_decinst_t_* dec_inst);
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// int16_t WebRtcG7221C_DecoderInit48(G722_1C_48_decinst_t_* dec_inst);
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//
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// int16_t WebRtcG7221C_Encode24(G722_1C_24_encinst_t_* enc_inst,
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// int16_t* input,
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// int16_t len,
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// int16_t* output);
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// int16_t WebRtcG7221C_Encode32(G722_1C_32_encinst_t_* enc_inst,
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// int16_t* input,
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// int16_t len,
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// int16_t* output);
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// int16_t WebRtcG7221C_Encode48(G722_1C_48_encinst_t_* enc_inst,
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// int16_t* input,
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// int16_t len,
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// int16_t* output);
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//
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// int16_t WebRtcG7221C_Decode24(G722_1C_24_decinst_t_* dec_inst,
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// int16_t* bitstream,
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// int16_t len,
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// int16_t* output);
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// int16_t WebRtcG7221C_Decode32(G722_1C_32_decinst_t_* dec_inst,
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// int16_t* bitstream,
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// int16_t len,
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// int16_t* output);
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// int16_t WebRtcG7221C_Decode48(G722_1C_48_decinst_t_* dec_inst,
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// int16_t* bitstream,
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// int16_t len,
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// int16_t* output);
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//
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// int16_t WebRtcG7221C_DecodePlc24(G722_1C_24_decinst_t_* dec_inst,
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// int16_t* output,
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// int16_t nr_lost_frames);
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// int16_t WebRtcG7221C_DecodePlc32(G722_1C_32_decinst_t_* dec_inst,
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// int16_t* output,
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// int16_t nr_lost_frames);
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// int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* dec_inst,
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// int16_t* output,
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// int16_t nr_lost_frames);
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#endif
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namespace webrtc {
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namespace acm2 {
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#ifndef WEBRTC_CODEC_G722_1C
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ACMG722_1C::ACMG722_1C(int16_t /* codec_id */)
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: operational_rate_(-1),
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encoder_inst_ptr_(NULL),
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encoder_inst_ptr_right_(NULL),
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encoder_inst24_ptr_(NULL),
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encoder_inst24_ptr_right_(NULL),
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encoder_inst32_ptr_(NULL),
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encoder_inst32_ptr_right_(NULL),
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encoder_inst48_ptr_(NULL),
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encoder_inst48_ptr_right_(NULL) {
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return;
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}
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ACMG722_1C::~ACMG722_1C() { return; }
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int16_t ACMG722_1C::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 ACMG722_1C::InternalInitEncoder(
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WebRtcACMCodecParams* /* codec_params */) {
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return -1;
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}
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ACMGenericCodec* ACMG722_1C::CreateInstance(void) { return NULL; }
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int16_t ACMG722_1C::InternalCreateEncoder() { return -1; }
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void ACMG722_1C::DestructEncoderSafe() { return; }
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void ACMG722_1C::InternalDestructEncoderInst(void* /* ptr_inst */) { return; }
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#else //===================== Actual Implementation =======================
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ACMG722_1C::ACMG722_1C(int16_t codec_id)
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: encoder_inst_ptr_(NULL),
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encoder_inst_ptr_right_(NULL),
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encoder_inst24_ptr_(NULL),
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encoder_inst24_ptr_right_(NULL),
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encoder_inst32_ptr_(NULL),
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encoder_inst32_ptr_right_(NULL),
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encoder_inst48_ptr_(NULL),
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encoder_inst48_ptr_right_(NULL) {
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codec_id_ = codec_id;
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if (codec_id_ == ACMCodecDB::kG722_1C_24) {
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operational_rate_ = 24000;
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} else if (codec_id_ == ACMCodecDB::kG722_1C_32) {
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operational_rate_ = 32000;
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} else if (codec_id_ == ACMCodecDB::kG722_1C_48) {
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operational_rate_ = 48000;
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} else {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"Wrong codec id for G722_1c.");
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operational_rate_ = -1;
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}
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return;
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}
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ACMG722_1C::~ACMG722_1C() {
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if (encoder_inst_ptr_ != NULL) {
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delete encoder_inst_ptr_;
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encoder_inst_ptr_ = NULL;
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}
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if (encoder_inst_ptr_right_ != NULL) {
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delete encoder_inst_ptr_right_;
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encoder_inst_ptr_right_ = NULL;
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}
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switch (operational_rate_) {
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case 24000: {
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encoder_inst24_ptr_ = NULL;
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encoder_inst24_ptr_right_ = NULL;
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break;
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}
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case 32000: {
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encoder_inst32_ptr_ = NULL;
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encoder_inst32_ptr_right_ = NULL;
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break;
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}
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case 48000: {
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encoder_inst48_ptr_ = NULL;
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encoder_inst48_ptr_right_ = NULL;
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break;
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}
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default: {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"Wrong rate for G722_1c.");
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break;
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}
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}
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return;
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}
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int16_t ACMG722_1C::InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) {
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int16_t left_channel[640];
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int16_t right_channel[640];
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int16_t len_in_bytes;
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int16_t out_bits[240];
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// If stereo, split input signal in left and right channel before encoding
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if (num_channels_ == 2) {
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for (int i = 0, j = 0; i < frame_len_smpl_ * 2; i += 2, j++) {
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left_channel[j] = in_audio_[in_audio_ix_read_ + i];
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right_channel[j] = in_audio_[in_audio_ix_read_ + i + 1];
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}
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} else {
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memcpy(left_channel, &in_audio_[in_audio_ix_read_], 640);
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}
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switch (operational_rate_) {
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case 24000: {
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len_in_bytes = WebRtcG7221C_Encode24(encoder_inst24_ptr_, left_channel,
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640, &out_bits[0]);
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if (num_channels_ == 2) {
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len_in_bytes += WebRtcG7221C_Encode24(encoder_inst24_ptr_right_,
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right_channel, 640,
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&out_bits[len_in_bytes / 2]);
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}
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break;
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}
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case 32000: {
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len_in_bytes = WebRtcG7221C_Encode32(encoder_inst32_ptr_, left_channel,
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640, &out_bits[0]);
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if (num_channels_ == 2) {
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len_in_bytes += WebRtcG7221C_Encode32(encoder_inst32_ptr_right_,
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right_channel, 640,
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&out_bits[len_in_bytes / 2]);
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}
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break;
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}
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case 48000: {
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len_in_bytes = WebRtcG7221C_Encode48(encoder_inst48_ptr_, left_channel,
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640, &out_bits[0]);
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if (num_channels_ == 2) {
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len_in_bytes += WebRtcG7221C_Encode48(encoder_inst48_ptr_right_,
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right_channel, 640,
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&out_bits[len_in_bytes / 2]);
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}
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break;
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}
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default: {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"InternalEncode: Wrong rate for G722_1c.");
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return -1;
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}
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}
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memcpy(bitstream, out_bits, len_in_bytes);
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*bitstream_len_byte = len_in_bytes;
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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_ += 640 * num_channels_;
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return *bitstream_len_byte;
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}
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int16_t ACMG722_1C::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
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int16_t ret;
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switch (operational_rate_) {
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case 24000: {
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ret = WebRtcG7221C_EncoderInit24(encoder_inst24_ptr_right_);
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if (ret < 0) {
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return ret;
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}
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return WebRtcG7221C_EncoderInit24(encoder_inst24_ptr_);
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}
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case 32000: {
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ret = WebRtcG7221C_EncoderInit32(encoder_inst32_ptr_right_);
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if (ret < 0) {
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return ret;
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}
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return WebRtcG7221C_EncoderInit32(encoder_inst32_ptr_);
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}
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case 48000: {
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ret = WebRtcG7221C_EncoderInit48(encoder_inst48_ptr_right_);
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if (ret < 0) {
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return ret;
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}
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return WebRtcG7221C_EncoderInit48(encoder_inst48_ptr_);
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}
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default: {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"InternalInitEncode: Wrong rate for G722_1c.");
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return -1;
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}
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}
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}
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ACMGenericCodec* ACMG722_1C::CreateInstance(void) { return NULL; }
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int16_t ACMG722_1C::InternalCreateEncoder() {
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if ((encoder_inst_ptr_ == NULL) || (encoder_inst_ptr_right_ == NULL)) {
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return -1;
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}
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switch (operational_rate_) {
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case 24000: {
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WebRtcG7221C_CreateEnc24(&encoder_inst24_ptr_);
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WebRtcG7221C_CreateEnc24(&encoder_inst24_ptr_right_);
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break;
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}
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case 32000: {
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WebRtcG7221C_CreateEnc32(&encoder_inst32_ptr_);
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WebRtcG7221C_CreateEnc32(&encoder_inst32_ptr_right_);
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break;
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}
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case 48000: {
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WebRtcG7221C_CreateEnc48(&encoder_inst48_ptr_);
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WebRtcG7221C_CreateEnc48(&encoder_inst48_ptr_right_);
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break;
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}
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default: {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
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"InternalCreateEncoder: Wrong rate for G722_1c.");
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return -1;
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}
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}
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return 0;
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}
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void ACMG722_1C::DestructEncoderSafe() {
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encoder_exist_ = false;
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encoder_initialized_ = false;
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if (encoder_inst_ptr_ != NULL) {
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delete encoder_inst_ptr_;
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encoder_inst_ptr_ = NULL;
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}
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if (encoder_inst_ptr_right_ != NULL) {
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delete encoder_inst_ptr_right_;
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encoder_inst_ptr_right_ = NULL;
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}
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encoder_inst24_ptr_ = NULL;
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encoder_inst32_ptr_ = NULL;
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encoder_inst48_ptr_ = NULL;
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}
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void ACMG722_1C::InternalDestructEncoderInst(void* ptr_inst) {
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if (ptr_inst != NULL) {
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delete 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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