bba7807078
This reverts portions of commitcb180976dd, which reverted commit83ad33a8ae. Specifically, the files in webrtc/modules/audio_coding/codecs/ that are not in ilbc/ or isac/, as well as webrtc/modules/audio_coding/main/test/opus_test.cc, are relanded. The original commit message is below: Upconvert various types to int. Per comments from HL/kwiberg on https://webrtc-codereview.appspot.com/42569004 , when there is existing usage of mixed types (int16_t, int, etc.), we'd prefer to standardize on larger types like int and phase out use of int16_t. Specifically, "Using int16 just because we're sure all reasonable values will fit in 16 bits isn't usually meaningful in C." This converts some existing uses of int16_t (and, in a few cases, other types such as uint16_t) to int (or, in a few places, int32_t). Other locations will be converted to size_t in a separate change. BUG=none TBR=kwiberg Review URL: https://codereview.webrtc.org/1179093003 Cr-Commit-Position: refs/heads/master@{#9424}
115 lines
3.0 KiB
C
115 lines
3.0 KiB
C
/*
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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 <stdlib.h>
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#include <string.h>
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#include "g722_enc_dec.h"
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#include "g722_interface.h"
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#include "webrtc/typedefs.h"
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int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
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{
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*G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
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if (*G722enc_inst!=NULL) {
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return(0);
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} else {
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return(-1);
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}
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}
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int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
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{
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// Create and/or reset the G.722 encoder
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// Bitrate 64 kbps and wideband mode (2)
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G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
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(G722EncoderState*) G722enc_inst, 64000, 2);
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if (G722enc_inst == NULL) {
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return -1;
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} else {
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return 0;
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}
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}
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int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
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{
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// Free encoder memory
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return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
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}
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int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
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const int16_t* speechIn,
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int16_t len,
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uint8_t* encoded)
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{
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unsigned char *codechar = (unsigned char*) encoded;
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// Encode the input speech vector
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return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
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speechIn, len);
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}
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int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
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{
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*G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
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if (*G722dec_inst!=NULL) {
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return(0);
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} else {
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return(-1);
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}
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}
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int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
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{
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// Create and/or reset the G.722 decoder
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// Bitrate 64 kbps and wideband mode (2)
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G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
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(G722DecoderState*) G722dec_inst, 64000, 2);
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if (G722dec_inst == NULL) {
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return -1;
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} else {
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return 0;
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}
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}
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int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
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{
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// Free encoder memory
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return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
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}
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int16_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
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const uint8_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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{
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// Decode the G.722 encoder stream
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*speechType=G722_WEBRTC_SPEECH;
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return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
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encoded, len);
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}
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int16_t WebRtcG722_Version(char *versionStr, short len)
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{
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// Get version string
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char version[30] = "2.0.0\n";
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if (strlen(version) < (unsigned int)len)
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{
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strcpy(versionStr, version);
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return 0;
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}
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else
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{
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return -1;
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}
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}
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