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Isolate AudioEncode submodule

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Isolate AudioEncode from rest of the project, minor cleanups
This commit is contained in:
Azamat H. Hackimov 2024-07-27 22:40:09 +03:00
parent 506521695a
commit f1737d2a19
3 changed files with 41 additions and 35 deletions
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15
AudioEncode/CMakeLists.txt
View File

@ -1,6 +1,13 @@
set(HEADERS)
set(CPPS
encoder.cpp)
encoder.cpp
)
add_library(AudioEncode STATIC ${HEADERS} ${CPPS})
target_link_libraries(AudioEncode libacm)
add_library(AudioEncode STATIC ${CPPS})
target_link_libraries(AudioEncode PRIVATE
libacm
)
target_include_directories(AudioEncode PUBLIC
$<BUILD_INTERFACE:
${PROJECT_SOURCE_DIR}/AudioEncode
>
)

13
lib/audio_encode.h → AudioEncode/audio_encode.h
View File

@ -1,5 +1,5 @@
/*
* Descent 3
* Descent 3
* Copyright (C) 2024 Parallax Software
*
* This program is free software: you can redistribute it and/or modify
@ -31,8 +31,8 @@
* $NoKeywords: $
*/
#ifndef __AUDIO_ENCODE_H_
#define __AUDIO_ENCODE_H_
#ifndef AUDIO_ENCODE_H_
#define AUDIO_ENCODE_H_
// input_levels (default 7 or for 2k total)
// input_samples (default 16 or for 2k total)
@ -40,8 +40,9 @@
// input_channels (default 1)
// input_factor (compression factor) (default 4 for 22K, 8 for 44K)
// input_volscale (Volume scaling) (slightly <= 1.0, default ,97)
bool aenc_Compress(char *input_filename, char *output_filename, int *input_levels = nullptr, int *input_samples = nullptr,
int *input_rate = nullptr, int *input_channels = nullptr, float *input_factor = nullptr,
float *input_volscale = nullptr);
bool aenc_Compress(char *input_filename, char *output_filename, const int *input_levels = nullptr,
const int *input_samples = nullptr, const int *input_rate = nullptr,
const int *input_channels = nullptr, const float *input_factor = nullptr,
const float *input_volscale = nullptr);
#endif

48
AudioEncode/encoder.cpp
View File

@ -1,25 +1,24 @@
/*
* Descent 3
* Copyright (C) 2024 Parallax Software
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
* Descent 3
* Copyright (C) 2024 Parallax Software
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include <cstdint>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <cstdio>
#include "audio_encode.h"
#include "mono.h"
#include "Aencode.h"
@ -37,8 +36,9 @@ int32_t aenc_ReadSamp(void *data) {
return (b << 8) | a;
}
bool aenc_Compress(char *input_filename, char *output_filename, int *input_levels, int *input_samples, int *input_rate,
int *input_channels, float *input_factor, float *input_volscale) {
bool aenc_Compress(char *input_filename, char *output_filename, const int *input_levels, const int *input_samples,
const int *input_rate, const int *input_channels, const float *input_factor,
const float *input_volscale) {
FILE *in, *out;
int32_t result;
@ -90,7 +90,7 @@ bool aenc_Compress(char *input_filename, char *output_filename, int *input_level
factor = *input_factor; // Factor of compression (default 4 for 22K, 8 for 44K)
factor_set = 1;
if (factor < 1.0f && factor)
if (factor != 0.0f && factor < 1.0f)
factor = 1.0f / factor;
if (factor <= 0.0f) {
@ -114,9 +114,7 @@ bool aenc_Compress(char *input_filename, char *output_filename, int *input_level
} else if (!levels_set) {
unsigned subbands = (2048 / samples_per_subband) >> 1;
for (levels = 0; subbands; subbands >>= 1, ++levels)
{
for (levels = 0; subbands; subbands >>= 1, ++levels) {
}
}