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godot/thirdparty/opus/silk/enc_API.c

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Add support for Opus audio format
2015-10-02 17:20:50 +00:00
/***********************************************************************
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.
***********************************************************************/
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 18:37:38 +00:00
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "define.h"
#include "API.h"
#include "control.h"
#include "typedef.h"
#include "stack_alloc.h"
#include "structs.h"
#include "tuning_parameters.h"
#ifdef FIXED_POINT
#include "main_FIX.h"
Add support for Opus audio format
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#else
ogg/vorbis/opus: Make them modules and unbundle thirdparty libs Took the opportunity to undo the Godot changed made to the opus source. The opus module should eventually be built in its own environment to avoid polluting others with too many include dirs and defines. TODO: Fix the platform/ stuff for opus.
2016-10-12 18:37:38 +00:00
#include "main_FLP.h"
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#endif
/***************************************/
/* Read control structure from encoder */
/***************************************/
static opus_int silk_QueryEncoder( /* O Returns error code */
const void *encState, /* I State */
silk_EncControlStruct *encStatus /* O Encoder Status */
);
/****************************************/
/* Encoder functions */
/****************************************/
opus_int silk_Get_Encoder_Size( /* O Returns error code */
opus_int *encSizeBytes /* O Number of bytes in SILK encoder state */
)
{
opus_int ret = SILK_NO_ERROR;
*encSizeBytes = sizeof( silk_encoder );
return ret;
}
/*************************/
/* Init or Reset encoder */
/*************************/
opus_int silk_InitEncoder( /* O Returns error code */
void *encState, /* I/O State */
int arch, /* I Run-time architecture */
silk_EncControlStruct *encStatus /* O Encoder Status */
)
{
silk_encoder *psEnc;
opus_int n, ret = SILK_NO_ERROR;
psEnc = (silk_encoder *)encState;
/* Reset encoder */
silk_memset( psEnc, 0, sizeof( silk_encoder ) );
for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) {
if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ], arch ) ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
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}
}
psEnc->nChannelsAPI = 1;
psEnc->nChannelsInternal = 1;
/* Read control structure */
if( ret += silk_QueryEncoder( encState, encStatus ) ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
Add support for Opus audio format
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}
return ret;
}
/***************************************/
/* Read control structure from encoder */
/***************************************/
static opus_int silk_QueryEncoder( /* O Returns error code */
const void *encState, /* I State */
silk_EncControlStruct *encStatus /* O Encoder Status */
)
{
opus_int ret = SILK_NO_ERROR;
silk_encoder_state_Fxx *state_Fxx;
silk_encoder *psEnc = (silk_encoder *)encState;
state_Fxx = psEnc->state_Fxx;
encStatus->nChannelsAPI = psEnc->nChannelsAPI;
encStatus->nChannelsInternal = psEnc->nChannelsInternal;
encStatus->API_sampleRate = state_Fxx[ 0 ].sCmn.API_fs_Hz;
encStatus->maxInternalSampleRate = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz;
encStatus->minInternalSampleRate = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz;
encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz;
encStatus->payloadSize_ms = state_Fxx[ 0 ].sCmn.PacketSize_ms;
encStatus->bitRate = state_Fxx[ 0 ].sCmn.TargetRate_bps;
encStatus->packetLossPercentage = state_Fxx[ 0 ].sCmn.PacketLoss_perc;
encStatus->complexity = state_Fxx[ 0 ].sCmn.Complexity;
encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC;
encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX;
encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR;
encStatus->internalSampleRate = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch;
encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0;
return ret;
}
/**************************/
/* Encode frame with Silk */
/**************************/
/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what */
/* encControl->payloadSize_ms is set to */
opus_int silk_Encode( /* O Returns error code */
void *encState, /* I/O State */
silk_EncControlStruct *encControl, /* I Control status */
const opus_int16 *samplesIn, /* I Speech sample input vector */
opus_int nSamplesIn, /* I Number of samples in input vector */
ec_enc *psRangeEnc, /* I/O Compressor data structure */
opus_int32 *nBytesOut, /* I/O Number of bytes in payload (input: Max bytes) */
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
const opus_int prefillFlag /* I Flag to indicate prefilling buffers no coding */
Add support for Opus audio format
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)
{
opus_int n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
opus_int nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms;
opus_int nSamplesFromInput = 0, nSamplesFromInputMax;
opus_int speech_act_thr_for_switch_Q8;
opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum;
silk_encoder *psEnc = ( silk_encoder * )encState;
VARDECL( opus_int16, buf );
opus_int transition, curr_block, tot_blocks;
SAVE_STACK;
if (encControl->reducedDependency)
{
psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1;
psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1;
}
psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
/* Check values in encoder control structure */
Update Opus driver to 1.1.2 And opusfile to 0.7.
2016-05-01 15:48:46 +00:00
if( ( ret = check_control_input( encControl ) ) != 0 ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
Add support for Opus audio format
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RESTORE_STACK;
return ret;
}
encControl->switchReady = 0;
if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) {
/* Mono -> Stereo transition: init state of second channel and stereo state */
ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ], psEnc->state_Fxx[ 0 ].sCmn.arch );
silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0;
psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1;
psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0;
psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1;
psEnc->sStereo.width_prev_Q14 = 0;
psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 );
if( psEnc->nChannelsAPI == 2 ) {
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) );
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State, &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State, sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) );
}
}
transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal);
psEnc->nChannelsAPI = encControl->nChannelsAPI;
psEnc->nChannelsInternal = encControl->nChannelsInternal;
nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1;
curr_block = 0;
if( prefillFlag ) {
/* Only accept input length of 10 ms */
if( nBlocksOf10ms != 1 ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
Add support for Opus audio format
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RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
/* Reset Encoder */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch );
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( !ret );
Add support for Opus audio format
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}
tmp_payloadSize_ms = encControl->payloadSize_ms;
encControl->payloadSize_ms = 10;
tmp_complexity = encControl->complexity;
encControl->complexity = 0;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1;
}
} else {
/* Only accept input lengths that are a multiple of 10 ms */
if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
Add support for Opus audio format
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RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
/* Make sure no more than one packet can be produced */
if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( 0 );
Add support for Opus audio format
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RESTORE_STACK;
return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
}
}
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 );
Add support for Opus audio format
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for( n = 0; n < encControl->nChannelsInternal; n++ ) {
/* Force the side channel to the same rate as the mid */
opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
Add support for Opus audio format
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silk_assert( 0 );
RESTORE_STACK;
return ret;
}
if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) {
for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0;
}
}
psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX;
}
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
Add support for Opus audio format
2015-10-02 17:20:50 +00:00
/* Input buffering/resampling and encoding */
nSamplesToBufferMax =
10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz;
nSamplesFromInputMax =
silk_DIV32_16( nSamplesToBufferMax *
psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz,
psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
ALLOC( buf, nSamplesFromInputMax, opus_int16 );
while( 1 ) {
nSamplesToBuffer = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
nSamplesToBuffer = silk_min( nSamplesToBuffer, nSamplesToBufferMax );
nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
/* Resample and write to buffer */
if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) {
opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
for( n = 0; n < nSamplesFromInput; n++ ) {
buf[ n ] = samplesIn[ 2 * n ];
}
/* Making sure to start both resamplers from the same state when switching from mono to stereo */
if( psEnc->nPrevChannelsInternal == 1 && id==0 ) {
silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state));
}
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
nSamplesToBuffer = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx;
nSamplesToBuffer = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
for( n = 0; n < nSamplesFromInput; n++ ) {
buf[ n ] = samplesIn[ 2 * n + 1 ];
}
ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer;
} else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) {
/* Combine left and right channels before resampling */
for( n = 0; n < nSamplesFromInput; n++ ) {
sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ];
buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
}
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
/* On the first mono frame, average the results for the two resampler states */
if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) {
ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) {
psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] =
silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ]
+ psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1);
}
}
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
} else {
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
Add support for Opus audio format
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silk_memcpy(buf, samplesIn, nSamplesFromInput*sizeof(opus_int16));
ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
&psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
}
samplesIn += nSamplesFromInput * encControl->nChannelsAPI;
nSamplesIn -= nSamplesFromInput;
/* Default */
psEnc->allowBandwidthSwitch = 0;
/* Silk encoder */
if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) {
/* Enough data in input buffer, so encode */
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length );
silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length );
Add support for Opus audio format
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/* Deal with LBRR data */
if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) {
/* Create space at start of payload for VAD and FEC flags */
opus_uint8 iCDF[ 2 ] = { 0, 0 };
iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
ec_enc_icdf( psRangeEnc, 0, iCDF, 8 );
/* Encode any LBRR data from previous packet */
/* Encode LBRR flags */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
LBRR_symbol = 0;
for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i );
}
psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0;
if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) {
ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 );
}
}
/* Code LBRR indices and excitation signals */
for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) {
opus_int condCoding;
if( encControl->nChannelsInternal == 2 && n == 0 ) {
silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] );
/* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */
if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) {
silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] );
}
}
/* Use conditional coding if previous frame available */
if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) {
condCoding = CODE_CONDITIONALLY;
} else {
condCoding = CODE_INDEPENDENTLY;
}
silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding );
silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType,
psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length );
}
}
}
/* Reset LBRR flags */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) );
}
Update Opus driver to 1.1.2 And opusfile to 0.7.
2016-05-01 15:48:46 +00:00
psEnc->nBitsUsedLBRR = ec_tell( psRangeEnc );
Add support for Opus audio format
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}
silk_HP_variable_cutoff( psEnc->state_Fxx );
/* Total target bits for packet */
nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
Update Opus driver to 1.1.2 And opusfile to 0.7.
2016-05-01 15:48:46 +00:00
/* Subtract bits used for LBRR */
Add support for Opus audio format
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if( !prefillFlag ) {
Update Opus driver to 1.1.2 And opusfile to 0.7.
2016-05-01 15:48:46 +00:00
nBits -= psEnc->nBitsUsedLBRR;
Add support for Opus audio format
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}
/* Divide by number of uncoded frames left in packet */
Update Opus driver to 1.1.2 And opusfile to 0.7.
2016-05-01 15:48:46 +00:00
nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket );
Add support for Opus audio format
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/* Convert to bits/second */
if( encControl->payloadSize_ms == 10 ) {
TargetRate_bps = silk_SMULBB( nBits, 100 );
} else {
TargetRate_bps = silk_SMULBB( nBits, 50 );
}
Update Opus driver to 1.1.2 And opusfile to 0.7.
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/* Subtract fraction of bits in excess of target in previous frames and packets */
Add support for Opus audio format
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TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
Update Opus driver to 1.1.2 And opusfile to 0.7.
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if( !prefillFlag && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded > 0 ) {
/* Compare actual vs target bits so far in this packet */
opus_int32 bitsBalance = ec_tell( psRangeEnc ) - psEnc->nBitsUsedLBRR - nBits * psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
TargetRate_bps -= silk_DIV32_16( silk_MUL( bitsBalance, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
}
Add support for Opus audio format
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/* Never exceed input bitrate */
TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 );
/* Convert Left/Right to Mid/Side */
if( encControl->nChannelsInternal == 2 ) {
silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ],
psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ],
MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono,
psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length );
if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
/* Reset side channel encoder memory for first frame with side coding */
if( psEnc->prev_decode_only_middle == 1 ) {
silk_memset( &psEnc->state_Fxx[ 1 ].sShape, 0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) );
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_memset( &psEnc->state_Fxx[ 1 ].sPrefilt, 0, sizeof( psEnc->state_Fxx[ 1 ].sPrefilt ) );
Add support for Opus audio format
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silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) );
silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) );
silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) );
psEnc->state_Fxx[ 1 ].sCmn.prevLag = 100;
psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev = 100;
psEnc->state_Fxx[ 1 ].sShape.LastGainIndex = 10;
psEnc->state_Fxx[ 1 ].sCmn.prevSignalType = TYPE_NO_VOICE_ACTIVITY;
psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_gain_Q16 = 65536;
psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1;
}
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ] );
Add support for Opus audio format
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} else {
psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0;
}
if( !prefillFlag ) {
silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
}
}
} else {
/* Buffering */
silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) );
silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) );
}
Revert "Update opus to 1.3.1 and opusfile to 0.11" This reverts commit e00426c512a7905f5f925d382c443bab7a0ca693. The way we handle platform-specific intrinsics is not good, so the current state will not compile on armv8. This commit also requires SSE4.1 support, which is likely not a good idea for portable binaries. We'll have to redo this with more caution after 3.2 is released, or we might simply drop opus as we're only using it as dependency for theora right now. Fixes #33606.
2019-11-18 08:56:18 +00:00
silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ] );
Add support for Opus audio format
2015-10-02 17:20:50 +00:00
/* Encode */
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
opus_int maxBits, useCBR;
/* Handling rate constraints */
maxBits = encControl->maxBits;
if( tot_blocks == 2 && curr_block == 0 ) {
maxBits = maxBits * 3 / 5;
} else if( tot_blocks == 3 ) {
if( curr_block == 0 ) {
maxBits = maxBits * 2 / 5;
} else if( curr_block == 1 ) {
maxBits = maxBits * 3 / 4;
}
}
useCBR = encControl->useCBR && curr_block == tot_blocks - 1;
if( encControl->nChannelsInternal == 1 ) {
channelRate_bps = TargetRate_bps;
} else {
channelRate_bps = MStargetRates_bps[ n ];
if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) {
useCBR = 0;
/* Give mid up to 1/2 of the max bits for that frame */
maxBits -= encControl->maxBits / ( tot_blocks * 2 );
}
}
if( channelRate_bps > 0 ) {
opus_int condCoding;
silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps );
/* Use independent coding if no previous frame available */
if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) {
condCoding = CODE_INDEPENDENTLY;
} else if( n > 0 && psEnc->prev_decode_only_middle ) {
/* If we skipped a side frame in this packet, we don't
need LTP scaling; the LTP state is well-defined. */
condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
} else {
condCoding = CODE_CONDITIONALLY;
}
if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) {
silk_assert( 0 );
}
}
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0;
psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++;
}
psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ];
/* Insert VAD and FEC flags at beginning of bitstream */
if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) {
flags = 0;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
flags = silk_LSHIFT( flags, 1 );
flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ];
}
flags = silk_LSHIFT( flags, 1 );
flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag;
}
if( !prefillFlag ) {
ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
}
/* Return zero bytes if all channels DTXed */
if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) {
*nBytesOut = 0;
}
psEnc->nBitsExceeded += *nBytesOut * 8;
psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
psEnc->nBitsExceeded = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 );
/* Update flag indicating if bandwidth switching is allowed */
speech_act_thr_for_switch_Q8 = silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ),
SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms );
if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) {
psEnc->allowBandwidthSwitch = 1;
psEnc->timeSinceSwitchAllowed_ms = 0;
} else {
psEnc->allowBandwidthSwitch = 0;
psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms;
}
}
if( nSamplesIn == 0 ) {
break;
}
} else {
break;
}
curr_block++;
}
psEnc->nPrevChannelsInternal = encControl->nChannelsInternal;
encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch;
encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0;
encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14;
if( prefillFlag ) {
encControl->payloadSize_ms = tmp_payloadSize_ms;
encControl->complexity = tmp_complexity;
for( n = 0; n < encControl->nChannelsInternal; n++ ) {
psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0;
}
}
RESTORE_STACK;
return ret;
}