session-android/jni/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

176 lines
4.2 KiB
C

/*
* Copyright (c) 2011 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 <string.h>
#include "g711.h"
#include "g711_interface.h"
#include "typedefs.h"
int16_t WebRtcG711_EncodeA(void* state,
int16_t* speechIn,
int16_t len,
int16_t* encoded) {
int n;
uint16_t tempVal, tempVal2;
// Set and discard to avoid getting warnings
(void)(state = NULL);
// Sanity check of input length
if (len < 0) {
return (-1);
}
// Loop over all samples
for (n = 0; n < len; n++) {
tempVal = (uint16_t) linear_to_alaw(speechIn[n]);
#ifdef WEBRTC_ARCH_BIG_ENDIAN
if ((n & 0x1) == 1) {
encoded[n >> 1] |= ((uint16_t) tempVal);
} else {
encoded[n >> 1] = ((uint16_t) tempVal) << 8;
}
#else
if ((n & 0x1) == 1) {
tempVal2 |= ((uint16_t) tempVal) << 8;
encoded[n >> 1] |= ((uint16_t) tempVal) << 8;
} else {
tempVal2 = ((uint16_t) tempVal);
encoded[n >> 1] = ((uint16_t) tempVal);
}
#endif
}
return (len);
}
int16_t WebRtcG711_EncodeU(void* state,
int16_t* speechIn,
int16_t len,
int16_t* encoded) {
int n;
uint16_t tempVal;
// Set and discard to avoid getting warnings
(void)(state = NULL);
// Sanity check of input length
if (len < 0) {
return (-1);
}
// Loop over all samples
for (n = 0; n < len; n++) {
tempVal = (uint16_t) linear_to_ulaw(speechIn[n]);
#ifdef WEBRTC_ARCH_BIG_ENDIAN
if ((n & 0x1) == 1) {
encoded[n >> 1] |= ((uint16_t) tempVal);
} else {
encoded[n >> 1] = ((uint16_t) tempVal) << 8;
}
#else
if ((n & 0x1) == 1) {
encoded[n >> 1] |= ((uint16_t) tempVal) << 8;
} else {
encoded[n >> 1] = ((uint16_t) tempVal);
}
#endif
}
return (len);
}
int16_t WebRtcG711_DecodeA(void* state,
int16_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType) {
int n;
uint16_t tempVal;
// Set and discard to avoid getting warnings
(void)(state = NULL);
// Sanity check of input length
if (len < 0) {
return (-1);
}
for (n = 0; n < len; n++) {
#ifdef WEBRTC_ARCH_BIG_ENDIAN
if ((n & 0x1) == 1) {
tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
} else {
tempVal = ((uint16_t) encoded[n >> 1] >> 8);
}
#else
if ((n & 0x1) == 1) {
tempVal = (encoded[n >> 1] >> 8);
} else {
tempVal = (encoded[n >> 1] & 0xFF);
}
#endif
decoded[n] = (int16_t) alaw_to_linear(tempVal);
}
*speechType = 1;
return (len);
}
int16_t WebRtcG711_DecodeU(void* state,
int16_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType) {
int n;
uint16_t tempVal;
// Set and discard to avoid getting warnings
(void)(state = NULL);
// Sanity check of input length
if (len < 0) {
return (-1);
}
for (n = 0; n < len; n++) {
#ifdef WEBRTC_ARCH_BIG_ENDIAN
if ((n & 0x1) == 1) {
tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
} else {
tempVal = ((uint16_t) encoded[n >> 1] >> 8);
}
#else
if ((n & 0x1) == 1) {
tempVal = (encoded[n >> 1] >> 8);
} else {
tempVal = (encoded[n >> 1] & 0xFF);
}
#endif
decoded[n] = (int16_t) ulaw_to_linear(tempVal);
}
*speechType = 1;
return (len);
}
int WebRtcG711_DurationEst(void* state,
const uint8_t* payload,
int payload_length_bytes) {
(void) state;
(void) payload;
/* G.711 is one byte per sample, so we can just return the number of bytes. */
return payload_length_bytes;
}
int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
strncpy(version, "2.0.0", lenBytes);
return 0;
}