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/*
Copyright (C) 2005-2009 Michel de Boer <michel@twinklephone.com>
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 2 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 <https://www.gnu.org/licenses/>.
*/
#include <cstdlib>
#include "audio_codecs.h"
unsigned short audio_sample_rate(t_audio_codec codec) {
switch(codec) {
case CODEC_G711_ALAW:
case CODEC_G711_ULAW:
case CODEC_GSM:
case CODEC_SPEEX_NB:
case CODEC_ILBC:
case CODEC_G729A:
case CODEC_G726_16:
case CODEC_G726_24:
case CODEC_G726_32:
case CODEC_G726_40:
case CODEC_TELEPHONE_EVENT:
return 8000;
case CODEC_SPEEX_WB:
return 16000;
case CODEC_SPEEX_UWB:
return 32000;
default:
// Use 8000 as default rate
return 8000;
}
}
bool is_speex_codec(t_audio_codec codec) {
return (codec == CODEC_SPEEX_NB ||
codec == CODEC_SPEEX_WB ||
codec == CODEC_SPEEX_UWB);
}
int resample(short *input_buf, int input_len, int input_sample_rate,
short *output_buf, int output_len, int output_sample_rate)
{
if (input_sample_rate > output_sample_rate) {
int downsample_factor = input_sample_rate / output_sample_rate;
int output_idx = -1;
for (int i = 0; i < input_len; i += downsample_factor) {
output_idx = i / downsample_factor;
if (output_idx >= output_len) {
// Output buffer is full
return output_len;
}
output_buf[output_idx] = input_buf[i];
}
return output_idx + 1;
} else {
int upsample_factor = output_sample_rate / input_sample_rate;
int output_idx = -1;
for (int i = 0; i < input_len; i++) {
for (int j = 0; j < upsample_factor; j++) {
output_idx = i * upsample_factor + j;
if (output_idx >= output_len) {
// Output buffer is full
return output_len;
}
output_buf[output_idx] = input_buf[i];
}
}
return output_idx + 1;
}
}
short mix_linear_pcm(short pcm1, short pcm2) {
long mixed_sample = long(pcm1) + long(pcm2);
// Compress a 17 bit PCM value into a 16-bit value.
// The easy way is to divide the value by 2, but this lowers
// the volume.
// Only lower the volume for the loud values. As for a normal
// voice call the values are not that loud, this gives better
// quality.
if (mixed_sample > 16384) {
mixed_sample = 16384 + (mixed_sample - 16384) / 3;
} else if (mixed_sample < -16384) {
mixed_sample = -16384 - (-16384 - mixed_sample) / 3;
}
return short(mixed_sample);
}
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