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/*
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* iaxclient: a cross-platform IAX softphone library
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*
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* Copyrights:
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* Copyright (C) 2003-2006, Horizon Wimba, Inc.
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* Copyright (C) 2007, Wimba, Inc.
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*
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* Contributors:
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* Steve Kann <stevek@stevek.com>
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* Michael Van Donselaar <mvand@vandonselaar.org>
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* Shawn Lawrence <shawn.lawrence@terracecomm.com>
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*
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* This program is free software, distributed under the terms of
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* the GNU Lesser (Library) General Public License.
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*/
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#include "audio_encode.h"
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#include "iaxclient_lib.h"
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#include "iax-client.h"
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#ifdef CODEC_GSM
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#include "codec_gsm.h"
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#endif
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#include "codec_ulaw.h"
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#include "codec_alaw.h"
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#include "codec_speex.h"
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#include <speex/speex_preprocess.h>
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#ifdef CODEC_ILBC
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#include "codec_ilbc.h"
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#endif
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float iaxci_silence_threshold = AUDIO_ENCODE_SILENCE_DB;
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static float input_level = 0.0f;
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static float output_level = 0.0f;
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static SpeexPreprocessState *st = NULL;
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static int speex_state_size = 0;
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static int speex_state_rate = 0;
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int iaxci_filters = IAXC_FILTER_AGC|IAXC_FILTER_DENOISE|IAXC_FILTER_AAGC|IAXC_FILTER_CN;
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/* use to measure time since last audio was processed */
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static struct timeval timeLastInput ;
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static struct timeval timeLastOutput ;
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static struct iaxc_speex_settings speex_settings =
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{
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1, /* decode_enhance */
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-1, /* float quality */
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-1, /* bitrate */
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0, /* vbr */
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0, /* abr */
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3 /* complexity */
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};
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static float vol_to_db(float vol)
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{
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/* avoid calling log10() on zero which yields inf or
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* negative numbers which yield nan */
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if ( vol <= 0.0f )
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return AUDIO_ENCODE_SILENCE_DB;
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else
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return log10f(vol) * 20.0f;
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}
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static int do_level_callback()
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{
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static struct timeval last = {0,0};
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struct timeval now;
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float input_db;
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float output_db;
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now = iax_tvnow();
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if ( last.tv_sec != 0 && iaxci_usecdiff(&now, &last) < 100000 )
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return 0;
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last = now;
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/* if input has not been processed in the last second, set to silent */
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input_db = iaxci_usecdiff(&now, &timeLastInput) < 1000000 ?
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vol_to_db(input_level) : AUDIO_ENCODE_SILENCE_DB;
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/* if output has not been processed in the last second, set to silent */
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output_db = iaxci_usecdiff(&now, &timeLastOutput) < 1000000 ?
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vol_to_db(output_level) : AUDIO_ENCODE_SILENCE_DB;
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iaxci_do_levels_callback(input_db, output_db);
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return 0;
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}
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static void set_speex_filters()
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{
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int i;
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if ( !st )
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return;
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i = 1; /* always make VAD decision */
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_VAD, &i);
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i = (iaxci_filters & IAXC_FILTER_AGC) ? 1 : 0;
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_AGC, &i);
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i = (iaxci_filters & IAXC_FILTER_DENOISE) ? 1 : 0;
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_DENOISE, &i);
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/*
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* We can tweak these parameters to play with VAD sensitivity.
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* For now, we use the default values since it seems they are a good starting point.
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* However, if need be, this is the code that needs to change
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*/
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i = 35;
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_PROB_START, &i);
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i = 20;
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_SET_PROB_CONTINUE, &i);
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}
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static void calculate_level(short *audio, int len, float *level)
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{
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int big_sample = 0;
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int i;
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for ( i = 0; i < len; i++ )
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{
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const int sample = abs(audio[i]);
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big_sample = sample > big_sample ?
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sample : big_sample;
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}
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*level += ((float)big_sample / 32767.0f - *level) / 5.0f;
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}
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static int input_postprocess(void *audio, int len, int rate)
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{
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static float lowest_volume = 1.0f;
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float volume;
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int silent = 0;
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if ( !st || speex_state_size != len || speex_state_rate != rate )
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{
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if (st)
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speex_preprocess_state_destroy(st);
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st = speex_preprocess_state_init(len,rate);
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speex_state_size = len;
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speex_state_rate = rate;
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set_speex_filters();
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}
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calculate_level((short *)audio, len, &input_level);
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/* only preprocess if we're interested in VAD, AGC, or DENOISE */
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if ( (iaxci_filters & (IAXC_FILTER_DENOISE | IAXC_FILTER_AGC)) ||
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iaxci_silence_threshold > 0.0f )
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silent = !speex_preprocess(st, (spx_int16_t *)audio, NULL);
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/* Analog AGC: Bring speex AGC gain out to mixer, with lots of hysteresis */
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/* use a higher continuation threshold for AAGC than for VAD itself */
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if ( !silent &&
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iaxci_silence_threshold != 0.0f &&
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(iaxci_filters & IAXC_FILTER_AGC) &&
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(iaxci_filters & IAXC_FILTER_AAGC)
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)
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{
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static int i = 0;
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i++;
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if ( (i & 0x3f) == 0 )
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{
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float loudness;
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#ifdef SPEEX_PREPROCESS_GET_AGC_LOUDNESS
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speex_preprocess_ctl(st, SPEEX_PREPROCESS_GET_AGC_LOUDNESS, &loudness);
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#else
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loudness = st->loudness2;
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#endif
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if ( loudness > 8000.0f || loudness < 4000.0f )
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{
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const float level = iaxc_input_level_get();
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if ( loudness > 16000.0f && level > 0.5f )
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{
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/* lower quickly if we're really too hot */
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iaxc_input_level_set(level - 0.2f);
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}
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else if ( loudness > 8000.0f && level >= 0.15f )
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{
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/* lower less quickly if we're a bit too hot */
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iaxc_input_level_set(level - 0.1f);
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}
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else if ( loudness < 4000.0f && level <= 0.9f )
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{
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/* raise slowly if we're cold */
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iaxc_input_level_set(level + 0.1f);
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}
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}
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}
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}
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/* This is ugly. Basically just don't get volume level if speex thought
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* we were silent. Just set it to 0 in that case */
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if ( iaxci_silence_threshold > 0.0f && silent )
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input_level = 0.0f;
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do_level_callback();
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volume = vol_to_db(input_level);
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if ( volume < lowest_volume )
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lowest_volume = volume;
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if ( iaxci_silence_threshold > 0.0f )
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return silent;
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else
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return volume < iaxci_silence_threshold;
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}
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static int output_postprocess(void *audio, int len)
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{
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calculate_level((short *)audio, len, &output_level);
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do_level_callback();
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return 0;
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}
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static struct iaxc_audio_codec *create_codec(int format)
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{
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switch (format & IAXC_AUDIO_FORMAT_MASK)
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{
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#ifdef CODEC_GSM
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case IAXC_FORMAT_GSM:
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return codec_audio_gsm_new();
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#endif
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case IAXC_FORMAT_ULAW:
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return codec_audio_ulaw_new();
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case IAXC_FORMAT_ALAW:
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return codec_audio_alaw_new();
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case IAXC_FORMAT_SPEEX:
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return codec_audio_speex_new(&speex_settings);
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#ifdef CODEC_ILBC
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case IAXC_FORMAT_ILBC:
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return codec_audio_ilbc_new();
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#endif
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default:
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/* ERROR: codec not supported */
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fprintf(stderr, "ERROR: Codec not supported: %d\n", format);
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return NULL;
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}
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}
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EXPORT void iaxc_set_speex_settings(int decode_enhance, float quality,
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int bitrate, int vbr, int abr, int complexity)
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{
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speex_settings.decode_enhance = decode_enhance;
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speex_settings.quality = quality;
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speex_settings.bitrate = bitrate;
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speex_settings.vbr = vbr;
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speex_settings.abr = abr;
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speex_settings.complexity = complexity;
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}
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int audio_send_encoded_audio(struct iaxc_call *call, int callNo, void *data,
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int format, int samples)
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{
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unsigned char outbuf[1024];
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int outsize = 1024;
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int silent;
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int insize = samples;
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/* update last input timestamp */
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timeLastInput = iax_tvnow();
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silent = input_postprocess(data, insize, 8000);
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if(silent)
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{
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if(!call->tx_silent)
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{ /* send a Comfort Noise Frame */
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call->tx_silent = 1;
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if ( iaxci_filters & IAXC_FILTER_CN )
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iax_send_cng(call->session, 10, NULL, 0);
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}
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return 0; /* poof! no encoding! */
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}
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/* we're going to send voice now */
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call->tx_silent = 0;
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/* destroy encoder if it is incorrect type */
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if(call->encoder && call->encoder->format != format)
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{
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call->encoder->destroy(call->encoder);
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call->encoder = NULL;
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}
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/* just break early if there's no format defined: this happens for the
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* first couple of frames of new calls */
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if(format == 0) return 0;
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/* create encoder if necessary */
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if(!call->encoder)
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{
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call->encoder = create_codec(format);
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}
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if(!call->encoder)
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{
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/* ERROR: no codec */
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fprintf(stderr, "ERROR: Codec could not be created: %d\n", format);
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return 0;
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}
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if(call->encoder->encode(call->encoder, &insize, (short *)data,
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&outsize, outbuf))
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{
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/* ERROR: codec error */
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fprintf(stderr, "ERROR: encode error: %d\n", format);
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return 0;
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}
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if(samples-insize == 0)
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{
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fprintf(stderr, "ERROR encoding (no samples output (samples=%d)\n", samples);
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return -1;
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}
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// Send the encoded audio data back to the app if required
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// TODO: fix the stupid way in which the encoded audio size is returned
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if ( iaxc_get_audio_prefs() & IAXC_AUDIO_PREF_RECV_LOCAL_ENCODED )
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iaxci_do_audio_callback(callNo, 0, IAXC_SOURCE_LOCAL, 1,
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call->encoder->format & IAXC_AUDIO_FORMAT_MASK,
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sizeof(outbuf) - outsize, outbuf);
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if(iax_send_voice(call->session,format, outbuf,
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sizeof(outbuf) - outsize, samples-insize) == -1)
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{
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fprintf(stderr, "Failed to send voice! %s\n", iax_errstr);
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return -1;
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}
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return 0;
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}
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/* decode encoded audio; return the number of bytes decoded
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* negative indicates error */
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int audio_decode_audio(struct iaxc_call * call, void * out, void * data, int len,
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int format, int * samples)
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{
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int insize = len;
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int outsize = *samples;
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timeLastOutput = iax_tvnow();
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if ( format == 0 )
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{
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fprintf(stderr, "audio_decode_audio: Format is zero (should't happen)!\n");
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return -1;
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}
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/* destroy decoder if it is incorrect type */
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if ( call->decoder && call->decoder->format != format )
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{
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call->decoder->destroy(call->decoder);
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call->decoder = NULL;
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}
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/* create decoder if necessary */
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if ( !call->decoder )
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{
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call->decoder = create_codec(format);
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}
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if ( !call->decoder )
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{
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fprintf(stderr, "ERROR: Codec could not be created: %d\n",
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format);
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return -1;
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}
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if ( call->decoder->decode(call->decoder,
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&insize, (unsigned char *)data,
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&outsize, (short *)out) )
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{
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fprintf(stderr, "ERROR: decode error: %d\n", format);
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return -1;
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}
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output_postprocess(out, *samples - outsize);
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*samples = outsize;
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return len - insize;
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}
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EXPORT int iaxc_get_filters(void)
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{
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return iaxci_filters;
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}
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EXPORT void iaxc_set_filters(int filters)
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{
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iaxci_filters = filters;
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set_speex_filters();
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}
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EXPORT void iaxc_set_silence_threshold(float thr)
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{
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iaxci_silence_threshold = thr;
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set_speex_filters();
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}
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