plus-messenger/TMessagesProj/jni/opus/silk/decoder_set_fs.c
DrKLO 4ede311055 Update to 1.4.6
Audio notes (opus codec)
A lot of different improvements and bug fixes

Thanks to:
https://github.com/DrKLO/Telegram/issues/293
https://github.com/DrKLO/Telegram/issues/256

FOSS configuration not ready yet

I will move main dev branch to github in next couple commits
2014-03-23 02:31:55 +04:00

109 lines
4.9 KiB
C

/***********************************************************************
Copyright (c) 2006-2011, Skype Limited. All rights reserved.
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
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- Redistributions of source code must retain the above copyright notice,
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names of specific contributors, may be used to endorse or promote
products derived from this software without specific prior written
permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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***********************************************************************/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "main.h"
/* Set decoder sampling rate */
opus_int silk_decoder_set_fs(
silk_decoder_state *psDec, /* I/O Decoder state pointer */
opus_int fs_kHz, /* I Sampling frequency (kHz) */
opus_int32 fs_API_Hz /* I API Sampling frequency (Hz) */
)
{
opus_int frame_length, ret = 0;
silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
silk_assert( psDec->nb_subfr == MAX_NB_SUBFR || psDec->nb_subfr == MAX_NB_SUBFR/2 );
/* New (sub)frame length */
psDec->subfr_length = silk_SMULBB( SUB_FRAME_LENGTH_MS, fs_kHz );
frame_length = silk_SMULBB( psDec->nb_subfr, psDec->subfr_length );
/* Initialize resampler when switching internal or external sampling frequency */
if( psDec->fs_kHz != fs_kHz || psDec->fs_API_hz != fs_API_Hz ) {
/* Initialize the resampler for dec_API.c preparing resampling from fs_kHz to API_fs_Hz */
ret += silk_resampler_init( &psDec->resampler_state, silk_SMULBB( fs_kHz, 1000 ), fs_API_Hz, 0 );
psDec->fs_API_hz = fs_API_Hz;
}
if( psDec->fs_kHz != fs_kHz || frame_length != psDec->frame_length ) {
if( fs_kHz == 8 ) {
if( psDec->nb_subfr == MAX_NB_SUBFR ) {
psDec->pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
} else {
psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
}
} else {
if( psDec->nb_subfr == MAX_NB_SUBFR ) {
psDec->pitch_contour_iCDF = silk_pitch_contour_iCDF;
} else {
psDec->pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
}
}
if( psDec->fs_kHz != fs_kHz ) {
psDec->ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
if( fs_kHz == 8 || fs_kHz == 12 ) {
psDec->LPC_order = MIN_LPC_ORDER;
psDec->psNLSF_CB = &silk_NLSF_CB_NB_MB;
} else {
psDec->LPC_order = MAX_LPC_ORDER;
psDec->psNLSF_CB = &silk_NLSF_CB_WB;
}
if( fs_kHz == 16 ) {
psDec->pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
} else if( fs_kHz == 12 ) {
psDec->pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
} else if( fs_kHz == 8 ) {
psDec->pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
} else {
/* unsupported sampling rate */
silk_assert( 0 );
}
psDec->first_frame_after_reset = 1;
psDec->lagPrev = 100;
psDec->LastGainIndex = 10;
psDec->prevSignalType = TYPE_NO_VOICE_ACTIVITY;
silk_memset( psDec->outBuf, 0, sizeof(psDec->outBuf));
silk_memset( psDec->sLPC_Q14_buf, 0, sizeof(psDec->sLPC_Q14_buf) );
}
psDec->fs_kHz = fs_kHz;
psDec->frame_length = frame_length;
}
/* Check that settings are valid */
silk_assert( psDec->frame_length > 0 && psDec->frame_length <= MAX_FRAME_LENGTH );
return ret;
}