c454fab03b
TEST=VoE auto-test, audio_coding_module_test; only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision Review URL: https://webrtc-codereview.appspot.com/937035 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
133 lines
4.2 KiB
C++
133 lines
4.2 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_pcmu.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 {
|
|
|
|
ACMPCMU::ACMPCMU(WebRtc_Word16 codec_id) {
|
|
codec_id_ = codec_id;
|
|
}
|
|
|
|
ACMPCMU::~ACMPCMU() {
|
|
return;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::InternalEncode(WebRtc_UWord8* bitstream,
|
|
WebRtc_Word16* bitstream_len_byte) {
|
|
*bitstream_len_byte = WebRtcG711_EncodeU(NULL, &in_audio_[in_audio_ix_read_],
|
|
frame_len_smpl_ * num_channels_,
|
|
(WebRtc_Word16*)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;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::DecodeSafe(WebRtc_UWord8* /* bitstream */,
|
|
WebRtc_Word16 /* bitstream_len_byte */,
|
|
WebRtc_Word16* /* audio */,
|
|
WebRtc_Word16* /* audio_samples */,
|
|
WebRtc_Word8* /* speech_type */) {
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::InternalInitEncoder(
|
|
WebRtcACMCodecParams* /* codec_params */) {
|
|
// This codec does not need initialization, PCM has no instance.
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::InternalInitDecoder(
|
|
WebRtcACMCodecParams* /* codec_params */) {
|
|
// This codec does not need initialization, PCM has no instance.
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word32 ACMPCMU::CodecDef(WebRtcNetEQ_CodecDef& codec_def,
|
|
const CodecInst& codec_inst) {
|
|
// Fill up the structure by calling
|
|
// "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
|
|
// Then call NetEQ to add the codec to it's database.
|
|
if (codec_inst.channels == 1) {
|
|
// Mono mode.
|
|
SET_CODEC_PAR(codec_def, kDecoderPCMu, codec_inst.pltype, NULL, 8000);
|
|
} else {
|
|
// Stereo mode.
|
|
SET_CODEC_PAR(codec_def, kDecoderPCMu_2ch, codec_inst.pltype, NULL, 8000);
|
|
}
|
|
SET_PCMU_FUNCTIONS(codec_def);
|
|
return 0;
|
|
}
|
|
|
|
ACMGenericCodec* ACMPCMU::CreateInstance(void) {
|
|
return NULL;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::InternalCreateEncoder() {
|
|
// PCM has no instance.
|
|
return 0;
|
|
}
|
|
|
|
WebRtc_Word16 ACMPCMU::InternalCreateDecoder() {
|
|
// PCM has no instance.
|
|
return 0;
|
|
}
|
|
|
|
void ACMPCMU::InternalDestructEncoderInst(void* /* ptr_inst */) {
|
|
// PCM has no instance.
|
|
return;
|
|
}
|
|
|
|
void ACMPCMU::DestructEncoderSafe() {
|
|
// PCM has no instance.
|
|
encoder_exist_ = false;
|
|
encoder_initialized_ = false;
|
|
return;
|
|
}
|
|
|
|
void ACMPCMU::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 ACMPCMU::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 webrtc
|