blob: e50d027dc87e6ece365fe0ddc724960982bf0d67 [file] [log] [blame]
/*
* 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_g7291.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 "webrtc/modules/audio_coding/main/codecs/g7291/interface/g7291_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_G729_1
ACMG729_1::ACMG729_1(int16_t /* codec_id */)
: encoder_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::InternalInitEncoder(
WebRtcACMCodecParams* /* codec_params */) {
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::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),
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;
}
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_],
reinterpret_cast<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::InternalInitEncoder(WebRtcACMCodecParams* codec_params) {
// set the bit rate and initialize
my_rate_ = codec_params->codec_inst.rate;
return SetBitRateSafe((uint32_t)my_rate_);
}
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::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 acm2
} // namespace webrtc