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Support for Signal calls.
Merge in RedPhone // FREEBIE
This commit is contained in:
175
jni/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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175
jni/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <string.h>
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#include "g711.h"
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#include "g711_interface.h"
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#include "typedefs.h"
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int16_t WebRtcG711_EncodeA(void* state,
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int16_t* speechIn,
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int16_t len,
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int16_t* encoded) {
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int n;
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uint16_t tempVal, tempVal2;
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// Set and discard to avoid getting warnings
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(void)(state = NULL);
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// Sanity check of input length
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if (len < 0) {
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return (-1);
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}
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// Loop over all samples
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for (n = 0; n < len; n++) {
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tempVal = (uint16_t) linear_to_alaw(speechIn[n]);
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#ifdef WEBRTC_ARCH_BIG_ENDIAN
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if ((n & 0x1) == 1) {
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encoded[n >> 1] |= ((uint16_t) tempVal);
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} else {
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encoded[n >> 1] = ((uint16_t) tempVal) << 8;
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}
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#else
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if ((n & 0x1) == 1) {
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tempVal2 |= ((uint16_t) tempVal) << 8;
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encoded[n >> 1] |= ((uint16_t) tempVal) << 8;
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} else {
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tempVal2 = ((uint16_t) tempVal);
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encoded[n >> 1] = ((uint16_t) tempVal);
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}
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#endif
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}
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return (len);
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}
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int16_t WebRtcG711_EncodeU(void* state,
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int16_t* speechIn,
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int16_t len,
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int16_t* encoded) {
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int n;
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uint16_t tempVal;
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// Set and discard to avoid getting warnings
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(void)(state = NULL);
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// Sanity check of input length
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if (len < 0) {
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return (-1);
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}
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// Loop over all samples
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for (n = 0; n < len; n++) {
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tempVal = (uint16_t) linear_to_ulaw(speechIn[n]);
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#ifdef WEBRTC_ARCH_BIG_ENDIAN
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if ((n & 0x1) == 1) {
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encoded[n >> 1] |= ((uint16_t) tempVal);
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} else {
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encoded[n >> 1] = ((uint16_t) tempVal) << 8;
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}
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#else
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if ((n & 0x1) == 1) {
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encoded[n >> 1] |= ((uint16_t) tempVal) << 8;
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} else {
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encoded[n >> 1] = ((uint16_t) tempVal);
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}
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#endif
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}
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return (len);
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}
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int16_t WebRtcG711_DecodeA(void* state,
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int16_t* encoded,
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int16_t len,
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int16_t* decoded,
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int16_t* speechType) {
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int n;
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uint16_t tempVal;
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// Set and discard to avoid getting warnings
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(void)(state = NULL);
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// Sanity check of input length
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if (len < 0) {
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return (-1);
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}
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for (n = 0; n < len; n++) {
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#ifdef WEBRTC_ARCH_BIG_ENDIAN
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if ((n & 0x1) == 1) {
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tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
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} else {
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tempVal = ((uint16_t) encoded[n >> 1] >> 8);
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}
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#else
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if ((n & 0x1) == 1) {
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tempVal = (encoded[n >> 1] >> 8);
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} else {
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tempVal = (encoded[n >> 1] & 0xFF);
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}
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#endif
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decoded[n] = (int16_t) alaw_to_linear(tempVal);
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}
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*speechType = 1;
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return (len);
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}
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int16_t WebRtcG711_DecodeU(void* state,
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int16_t* encoded,
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int16_t len,
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int16_t* decoded,
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int16_t* speechType) {
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int n;
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uint16_t tempVal;
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// Set and discard to avoid getting warnings
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(void)(state = NULL);
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// Sanity check of input length
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if (len < 0) {
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return (-1);
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}
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for (n = 0; n < len; n++) {
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#ifdef WEBRTC_ARCH_BIG_ENDIAN
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if ((n & 0x1) == 1) {
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tempVal = ((uint16_t) encoded[n >> 1] & 0xFF);
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} else {
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tempVal = ((uint16_t) encoded[n >> 1] >> 8);
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}
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#else
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if ((n & 0x1) == 1) {
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tempVal = (encoded[n >> 1] >> 8);
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} else {
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tempVal = (encoded[n >> 1] & 0xFF);
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}
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#endif
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decoded[n] = (int16_t) ulaw_to_linear(tempVal);
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}
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*speechType = 1;
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return (len);
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}
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int WebRtcG711_DurationEst(void* state,
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const uint8_t* payload,
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int payload_length_bytes) {
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(void) state;
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(void) payload;
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/* G.711 is one byte per sample, so we can just return the number of bytes. */
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return payload_length_bytes;
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}
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int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
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strncpy(version, "2.0.0", lenBytes);
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return 0;
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}
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