blob: c64641771ce7a7a2cf813b59a7817fb435c26741 [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/source/acm_pcma.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.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"
// Codec interface
namespace webrtc {
namespace acm1 {
ACMPCMA::ACMPCMA(int16_t codec_id) {
codec_id_ = codec_id;
}
ACMPCMA::~ACMPCMA() {
return;
}
int16_t ACMPCMA::InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte) {
*bitstream_len_byte = WebRtcG711_EncodeA(NULL, &in_audio_[in_audio_ix_read_],
frame_len_smpl_ * num_channels_,
(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_ * num_channels_;
return *bitstream_len_byte;
}
int16_t ACMPCMA::DecodeSafe(uint8_t* /* bitstream */,
int16_t /* bitstream_len_byte */,
int16_t* /* audio */,
int16_t* /* audio_samples */,
int8_t* /* speech_type */) {
return 0;
}
int16_t ACMPCMA::InternalInitEncoder(
WebRtcACMCodecParams* /* codec_params */) {
// This codec does not need initialization, PCM has no instance.
return 0;
}
int16_t ACMPCMA::InternalInitDecoder(
WebRtcACMCodecParams* /* codec_params */) {
// This codec does not need initialization, PCM has no instance.
return 0;
}
int32_t ACMPCMA::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
const CodecInst& codec_inst) {
// Fill up the structure by calling
// "SET_CODEC_PAR" & "SET_PCMA_FUNCTION."
// Then call NetEQ to add the codec to it's database.
if (codec_inst.channels == 1) {
// Mono mode.
SET_CODEC_PAR(codec_def, kDecoderPCMa, codec_inst.pltype, NULL, 8000);
} else {
// Stereo mode.
SET_CODEC_PAR(codec_def, kDecoderPCMa_2ch, codec_inst.pltype, NULL, 8000);
}
SET_PCMA_FUNCTIONS(codec_def);
return 0;
}
ACMGenericCodec* ACMPCMA::CreateInstance(void) {
return NULL;
}
int16_t ACMPCMA::InternalCreateEncoder() {
// PCM has no instance.
return 0;
}
int16_t ACMPCMA::InternalCreateDecoder() {
// PCM has no instance.
return 0;
}
void ACMPCMA::InternalDestructEncoderInst(void* /* ptr_inst */) {
// PCM has no instance.
return;
}
void ACMPCMA::DestructEncoderSafe() {
// PCM has no instance.
return;
}
void ACMPCMA::DestructDecoderSafe() {
// PCM has no instance.
decoder_initialized_ = false;
decoder_exist_ = false;
return;
}
// Split the stereo packet and place left and right channel after each other
// in the payload vector.
void ACMPCMA::SplitStereoPacket(uint8_t* payload, int32_t* payload_length) {
uint8_t right_byte;
// Check for valid inputs.
assert(payload != NULL);
assert(*payload_length > 0);
// Move one bytes representing right channel each loop, and place it at the
// end of the bytestream vector. After looping the data is reordered to:
// l1 l2 l3 l4 ... l(N-1) lN r1 r2 r3 r4 ... r(N-1) r(N),
// where N is the total number of samples.
for (int i = 0; i < *payload_length / 2; i++) {
right_byte = payload[i + 1];
memmove(&payload[i + 1], &payload[i + 2], *payload_length - i - 2);
payload[*payload_length - 1] = right_byte;
}
}
} // namespace acm1
} // namespace webrtc