Updating Opus to a pre-release of 1.1

silk/enc_API.c

 /***********************************************************************
 Copyright (c) 2006-2011, Skype Limited. All rights reserved.
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 modification, are permitted provided that the following conditions
 are met:
 - Redistributions of source code must retain the above copyright notice,
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 - Redistributions in binary form must reproduce the above copyright
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-- Neither the name of Internet Society, IETF or IETF Trust, nor the 
+- 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”
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
 AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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 ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
 LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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 #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"
 #else
 #include "main_FLP.h"
 #endif
 
 /***************************************/
 /* Read control structure from encoder */
@@ skipping 94 matching lines @@
     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)   */
     const opus_int                  prefillFlag         /* I    Flag to indicate prefilling buffers no coding   */
 )
 {
     opus_int   n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
-    opus_int   nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 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;
-    opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ];
+    VARDECL( opus_int16, buf );
     opus_int transition, curr_block, tot_blocks;
+    SAVE_STACK;
 
     psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
 
     /* Check values in encoder control structure */
     if( ( ret = check_control_input( encControl ) != 0 ) ) {
         silk_assert( 0 );
+        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 ] );
         silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
         silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
@@ skipping 13 matching lines @@
 
     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 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
         }
         /* Reset Encoder */
         for( n = 0; n < encControl->nChannelsInternal; n++ ) {
             if( (ret = silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) != 0 ) {
                 silk_assert( 0 );
             }
         }
         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 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            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 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
         }
     }
 
     TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 );
     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;
         if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
             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;
     }
     silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
 
     /* 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, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz );
+        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));
@@ skipping 268 matching lines @@
     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;
 }