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Diffstat (limited to 'src/audio/audio_codecs.h')
| -rw-r--r-- | src/audio/audio_codecs.h | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/src/audio/audio_codecs.h b/src/audio/audio_codecs.h new file mode 100644 index 0000000..54967b1 --- /dev/null +++ b/src/audio/audio_codecs.h @@ -0,0 +1,107 @@ +/* + 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, write to the Free Software + Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +*/ + +#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 +}; + +// 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 |