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-rw-r--r--media/libopus/silk/control_audio_bandwidth.c126
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diff --git a/media/libopus/silk/control_audio_bandwidth.c b/media/libopus/silk/control_audio_bandwidth.c
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+++ b/media/libopus/silk/control_audio_bandwidth.c
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+/***********************************************************************
+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
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+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
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "main.h"
+#include "tuning_parameters.h"
+
+/* Control internal sampling rate */
+opus_int silk_control_audio_bandwidth(
+ silk_encoder_state *psEncC, /* I/O Pointer to Silk encoder state */
+ silk_EncControlStruct *encControl /* I Control structure */
+)
+{
+ opus_int fs_kHz;
+ opus_int32 fs_Hz;
+
+ fs_kHz = psEncC->fs_kHz;
+ fs_Hz = silk_SMULBB( fs_kHz, 1000 );
+ if( fs_Hz == 0 ) {
+ /* Encoder has just been initialized */
+ fs_Hz = silk_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz );
+ fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+ } else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) {
+ /* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
+ fs_Hz = psEncC->API_fs_Hz;
+ fs_Hz = silk_min( fs_Hz, psEncC->maxInternal_fs_Hz );
+ fs_Hz = silk_max( fs_Hz, psEncC->minInternal_fs_Hz );
+ fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+ } else {
+ /* State machine for the internal sampling rate switching */
+ if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) {
+ /* Stop transition phase */
+ psEncC->sLP.mode = 0;
+ }
+ if( psEncC->allow_bandwidth_switch || encControl->opusCanSwitch ) {
+ /* Check if we should switch down */
+ if( silk_SMULBB( psEncC->fs_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz )
+ {
+ /* Switch down */
+ if( psEncC->sLP.mode == 0 ) {
+ /* New transition */
+ psEncC->sLP.transition_frame_no = TRANSITION_FRAMES;
+
+ /* Reset transition filter state */
+ silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+ }
+ if( encControl->opusCanSwitch ) {
+ /* Stop transition phase */
+ psEncC->sLP.mode = 0;
+
+ /* Switch to a lower sample frequency */
+ fs_kHz = psEncC->fs_kHz == 16 ? 12 : 8;
+ } else {
+ if( psEncC->sLP.transition_frame_no <= 0 ) {
+ encControl->switchReady = 1;
+ /* Make room for redundancy */
+ encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+ } else {
+ /* Direction: down (at double speed) */
+ psEncC->sLP.mode = -2;
+ }
+ }
+ }
+ else
+ /* Check if we should switch up */
+ if( silk_SMULBB( psEncC->fs_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz )
+ {
+ /* Switch up */
+ if( encControl->opusCanSwitch ) {
+ /* Switch to a higher sample frequency */
+ fs_kHz = psEncC->fs_kHz == 8 ? 12 : 16;
+
+ /* New transition */
+ psEncC->sLP.transition_frame_no = 0;
+
+ /* Reset transition filter state */
+ silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+
+ /* Direction: up */
+ psEncC->sLP.mode = 1;
+ } else {
+ if( psEncC->sLP.mode == 0 ) {
+ encControl->switchReady = 1;
+ /* Make room for redundancy */
+ encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+ } else {
+ /* Direction: up */
+ psEncC->sLP.mode = 1;
+ }
+ }
+ } else {
+ if (psEncC->sLP.mode<0)
+ psEncC->sLP.mode = 1;
+ }
+ }
+ }
+
+ return fs_kHz;
+}