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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/>.
*/
#ifndef _AUDIO_CODECS_H
#define _AUDIO_CODECS_H
#include "g711.h"
#include "g72x.h"
// Audio codecs
enum t_audio_codec {
CODEC_NULL,
CODEC_UNSUPPORTED,
CODEC_G711_ALAW,
CODEC_G711_ULAW,
CODEC_GSM,
CODEC_SPEEX_NB,
CODEC_SPEEX_WB,
CODEC_SPEEX_UWB,
CODEC_ILBC,
CODEC_G726_16,
CODEC_G726_24,
CODEC_G726_32,
CODEC_G726_40,
CODEC_TELEPHONE_EVENT,
CODEC_G729A
};
// Default ptime values (ms) for audio codecs
#define PTIME_G711_ALAW 20
#define PTIME_G711_ULAW 20
#define PTIME_G726 20
#define PTIME_GSM 20
#define PTIME_SPEEX 20
#define MIN_PTIME 10
#define MAX_PTIME 80
// Audio sample settings
#define AUDIO_SAMPLE_SIZE 16
// Maximum length (in packets) for concealment of lost packets
#define MAX_CONCEALMENT 2
// Size of jitter buffer in ms
// The jitter buffer is used to smooth playing out incoming RTP packets.
// The size of the buffer is also used as the expiry time in the ccRTP
// stack. Packets that have timestamp that is older than then size of
// the jitter buffer will not be sent out anymore.
#define JITTER_BUF_MS 80
// Duration of the expiry timer in the RTP stack.
// The ccRTP stack checks all data delivered to it against its clock.
// If the data is too old it will not send it out. Data can be old
// for several reasons:
//
// 1) The thread reading the soundcard has been paused for a while
// 2) The audio card buffers sound before releasing it.
//
// Especially the latter seems to happen on some soundcards. Data
// not older than defined delay are still allowed to go out. It's up
// to the receiving and to deal with the jitter this may cause.
#define MAX_OUT_AUDIO_DELAY_MS 160
// Buffer sizes
#define JITTER_BUF_SIZE(sample_rate) (JITTER_BUF_MS * (sample_rate)/1000 * AUDIO_SAMPLE_SIZE/8)
// Log speex errors
#define LOG_SPEEX_ERROR(func, spxfunc, spxerr) {\
log_file->write_header((func), LOG_NORMAL, LOG_DEBUG);\
log_file->write_raw("Speex error: ");\
log_file->write_raw((spxfunc));\
log_file->write_raw(" returned ");\
log_file->write_raw((spxerr));\
log_file->write_footer(); }
// Return the sampling rate for a codec
unsigned short audio_sample_rate(t_audio_codec codec);
// Returns true if the codec is a speex codec
bool is_speex_codec(t_audio_codec codec);
// Resample the input buffer to the output buffer
// Returns the number of samples put in the output buffer
// If the output buffer is too small, the number of samples will be
// truncated.
int resample(short *input_buf, int input_len, int input_sample_rate,
short *output_buf, int output_len, int output_sample_rate);
// Mix 2 16 bits signed linear PCM values
short mix_linear_pcm(short pcm1, short pcm2);
#endif
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