0946a56023
BUG=314 Review URL: https://webrtc-codereview.appspot.com/1271006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
346 lines
9.1 KiB
C++
346 lines
9.1 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/source/acm_g7291.h"
|
|
|
|
#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
|
|
#include "webrtc/modules/audio_coding/main/source/acm_neteq.h"
|
|
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
|
|
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
|
|
#include "webrtc/system_wrappers/interface/trace.h"
|
|
#ifdef WEBRTC_CODEC_G729_1
|
|
// NOTE! G.729.1 is not included in the open-source package. Modify this file
|
|
// or your codec API to match the function calls and names of used G.729.1 API
|
|
// file.
|
|
#include "g7291_interface.h"
|
|
#endif
|
|
|
|
namespace webrtc {
|
|
|
|
#ifndef WEBRTC_CODEC_G729_1
|
|
|
|
ACMG729_1::ACMG729_1(int16_t /* codec_id */)
|
|
: encoder_inst_ptr_(NULL),
|
|
decoder_inst_ptr_(NULL),
|
|
my_rate_(32000),
|
|
flag_8khz_(0),
|
|
flag_g729_mode_(0) {
|
|
return;
|
|
}
|
|
|
|
ACMG729_1::~ACMG729_1() {
|
|
return;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalEncode(
|
|
uint8_t* /* bitstream */,
|
|
int16_t* /* bitstream_len_byte */) {
|
|
return -1;
|
|
}
|
|
|
|
int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
|
|
int16_t /* bitstream_len_byte */,
|
|
int16_t* /* audio */,
|
|
int16_t* /* audio_samples */,
|
|
int8_t* /* speech_type */) {
|
|
return -1;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalInitEncoder(
|
|
WebRtcACMCodecParams* /* codec_params */) {
|
|
return -1;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalInitDecoder(
|
|
WebRtcACMCodecParams* /* codec_params */) {
|
|
return -1;
|
|
}
|
|
|
|
int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& /* codec_def */,
|
|
const CodecInst& /* codec_inst */) {
|
|
return -1;
|
|
}
|
|
|
|
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
|
|
return NULL;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalCreateEncoder() {
|
|
return -1;
|
|
}
|
|
|
|
void ACMG729_1::DestructEncoderSafe() {
|
|
return;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalCreateDecoder() {
|
|
return -1;
|
|
}
|
|
|
|
void ACMG729_1::DestructDecoderSafe() {
|
|
return;
|
|
}
|
|
|
|
void ACMG729_1::InternalDestructEncoderInst(void* /* ptr_inst */) {
|
|
return;
|
|
}
|
|
|
|
int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) {
|
|
return -1;
|
|
}
|
|
|
|
#else //===================== Actual Implementation =======================
|
|
|
|
struct G729_1_inst_t_;
|
|
|
|
ACMG729_1::ACMG729_1(int16_t codec_id)
|
|
: encoder_inst_ptr_(NULL),
|
|
decoder_inst_ptr_(NULL),
|
|
my_rate_(32000), // Default rate.
|
|
flag_8khz_(0),
|
|
flag_g729_mode_(0) {
|
|
// TODO(tlegrand): We should add codec_id as a input variable to the
|
|
// constructor of ACMGenericCodec.
|
|
codec_id_ = codec_id;
|
|
return;
|
|
}
|
|
|
|
ACMG729_1::~ACMG729_1() {
|
|
if (encoder_inst_ptr_ != NULL) {
|
|
WebRtcG7291_Free(encoder_inst_ptr_);
|
|
encoder_inst_ptr_ = NULL;
|
|
}
|
|
if (decoder_inst_ptr_ != NULL) {
|
|
WebRtcG7291_Free(decoder_inst_ptr_);
|
|
decoder_inst_ptr_ = NULL;
|
|
}
|
|
return;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalEncode(uint8_t* bitstream,
|
|
int16_t* bitstream_len_byte) {
|
|
|
|
// Initialize before entering the loop
|
|
int16_t num_encoded_samples = 0;
|
|
*bitstream_len_byte = 0;
|
|
|
|
int16_t byte_length_frame = 0;
|
|
|
|
// Derive number of 20ms frames per encoded packet.
|
|
// [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
|
|
int16_t num_20ms_frames = (frame_len_smpl_ / 320);
|
|
// Byte length for the frame. +1 is for rate information.
|
|
byte_length_frame = my_rate_ / (8 * 50) * num_20ms_frames + (1 -
|
|
flag_g729_mode_);
|
|
|
|
// The following might be revised if we have G729.1 Annex C (support for DTX);
|
|
do {
|
|
*bitstream_len_byte = WebRtcG7291_Encode(encoder_inst_ptr_,
|
|
&in_audio_[in_audio_ix_read_],
|
|
(int16_t*) bitstream,
|
|
my_rate_, num_20ms_frames);
|
|
|
|
// increment the read index this tell the caller that how far
|
|
// we have gone forward in reading the audio buffer
|
|
in_audio_ix_read_ += 160;
|
|
|
|
// sanity check
|
|
if (*bitstream_len_byte < 0) {
|
|
// error has happened
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"InternalEncode: Encode error for G729_1");
|
|
*bitstream_len_byte = 0;
|
|
return -1;
|
|
}
|
|
|
|
num_encoded_samples += 160;
|
|
} while (*bitstream_len_byte == 0);
|
|
|
|
// This criteria will change if we have Annex C.
|
|
if (*bitstream_len_byte != byte_length_frame) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"InternalEncode: Encode error for G729_1");
|
|
*bitstream_len_byte = 0;
|
|
return -1;
|
|
}
|
|
|
|
if (num_encoded_samples != frame_len_smpl_) {
|
|
*bitstream_len_byte = 0;
|
|
return -1;
|
|
}
|
|
|
|
return *bitstream_len_byte;
|
|
}
|
|
|
|
int16_t ACMG729_1::DecodeSafe(uint8_t* /* bitstream */,
|
|
int16_t /* bitstream_len_byte */,
|
|
int16_t* /* audio */,
|
|
int16_t* /* audio_samples */,
|
|
int8_t* /* speech_type */) {
|
|
return 0;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalInitEncoder(
|
|
WebRtcACMCodecParams* codec_params) {
|
|
//set the bit rate and initialize
|
|
my_rate_ = codec_params->codec_inst.rate;
|
|
return SetBitRateSafe((uint32_t) my_rate_);
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalInitDecoder(
|
|
WebRtcACMCodecParams* /* codec_params */) {
|
|
if (WebRtcG7291_DecoderInit(decoder_inst_ptr_) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"InternalInitDecoder: init decoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ACMG729_1::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
|
|
const CodecInst& codec_inst) {
|
|
if (!decoder_initialized_) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"CodeDef: Decoder uninitialized for G729_1");
|
|
return -1;
|
|
}
|
|
|
|
// Fill up the structure by calling
|
|
// "SET_CODEC_PAR" & "SET_G729_FUNCTION."
|
|
// Then call NetEQ to add the codec to it's
|
|
// database.
|
|
SET_CODEC_PAR((codec_def), kDecoderG729_1, codec_inst.pltype,
|
|
decoder_inst_ptr_, 16000);
|
|
SET_G729_1_FUNCTIONS((codec_def));
|
|
return 0;
|
|
}
|
|
|
|
ACMGenericCodec* ACMG729_1::CreateInstance(void) {
|
|
return NULL;
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalCreateEncoder() {
|
|
if (WebRtcG7291_Create(&encoder_inst_ptr_) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"InternalCreateEncoder: create encoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ACMG729_1::DestructEncoderSafe() {
|
|
encoder_exist_ = false;
|
|
encoder_initialized_ = false;
|
|
if (encoder_inst_ptr_ != NULL) {
|
|
WebRtcG7291_Free(encoder_inst_ptr_);
|
|
encoder_inst_ptr_ = NULL;
|
|
}
|
|
}
|
|
|
|
int16_t ACMG729_1::InternalCreateDecoder() {
|
|
if (WebRtcG7291_Create(&decoder_inst_ptr_) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"InternalCreateDecoder: create decoder failed for G729_1");
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ACMG729_1::DestructDecoderSafe() {
|
|
decoder_exist_ = false;
|
|
decoder_initialized_ = false;
|
|
if (decoder_inst_ptr_ != NULL) {
|
|
WebRtcG7291_Free(decoder_inst_ptr_);
|
|
decoder_inst_ptr_ = NULL;
|
|
}
|
|
}
|
|
|
|
void ACMG729_1::InternalDestructEncoderInst(void* ptr_inst) {
|
|
if (ptr_inst != NULL) {
|
|
// WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
|
|
}
|
|
return;
|
|
}
|
|
|
|
int16_t ACMG729_1::SetBitRateSafe(const int32_t rate) {
|
|
// allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
|
|
// 22000, 24000, 26000, 28000, 30000, 32000};
|
|
// TODO(tlegrand): This check exists in one other place two. Should be
|
|
// possible to reuse code.
|
|
switch (rate) {
|
|
case 8000: {
|
|
my_rate_ = 8000;
|
|
break;
|
|
}
|
|
case 12000: {
|
|
my_rate_ = 12000;
|
|
break;
|
|
}
|
|
case 14000: {
|
|
my_rate_ = 14000;
|
|
break;
|
|
}
|
|
case 16000: {
|
|
my_rate_ = 16000;
|
|
break;
|
|
}
|
|
case 18000: {
|
|
my_rate_ = 18000;
|
|
break;
|
|
}
|
|
case 20000: {
|
|
my_rate_ = 20000;
|
|
break;
|
|
}
|
|
case 22000: {
|
|
my_rate_ = 22000;
|
|
break;
|
|
}
|
|
case 24000: {
|
|
my_rate_ = 24000;
|
|
break;
|
|
}
|
|
case 26000: {
|
|
my_rate_ = 26000;
|
|
break;
|
|
}
|
|
case 28000: {
|
|
my_rate_ = 28000;
|
|
break;
|
|
}
|
|
case 30000: {
|
|
my_rate_ = 30000;
|
|
break;
|
|
}
|
|
case 32000: {
|
|
my_rate_ = 32000;
|
|
break;
|
|
}
|
|
default: {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, unique_id_,
|
|
"SetBitRateSafe: Invalid rate G729_1");
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
// Re-init with new rate
|
|
if (WebRtcG7291_EncoderInit(encoder_inst_ptr_, my_rate_, flag_8khz_,
|
|
flag_g729_mode_) >= 0) {
|
|
encoder_params_.codec_inst.rate = my_rate_;
|
|
return 0;
|
|
} else {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
#endif
|
|
|
|
} // namespace webrtc
|