Untangle fft third party lib from dependon WebRTC
TBR=phoglund Bug: webrtc:9558 Change-Id: I6cc1936549f008694c3617c1d990524c34da16e3 Reviewed-on: https://webrtc-review.googlesource.com/90411 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24115}
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@ -18,6 +18,7 @@
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#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
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#include "modules/audio_coding/codecs/isac/fix/source/fft.h"
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#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
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#include "modules/third_party/fft/fft.h"
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/* Tables are defined in transform_tables.c file or ARM assembly files. */
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/* Cosine table 1 in Q14 */
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@ -22,6 +22,7 @@
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#include <stddef.h>
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#include "modules/audio_coding/codecs/isac/main/source/structs.h"
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#include "modules/third_party/fft/fft.h"
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void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
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@ -24,9 +24,6 @@
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/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
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#define INITIAL_FRAMESAMPLES 960
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#define MAXFFTSIZE 2048
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#define NFACTOR 11
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/* do not modify the following; this will have to be modified if we
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* have a 20ms framesize option */
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/**********************************************************************/
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@ -20,6 +20,7 @@
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#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
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#include "modules/audio_coding/codecs/isac/main/source/settings.h"
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#include "modules/third_party/fft/fft.h"
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typedef struct Bitstreamstruct {
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uint8_t stream[STREAM_SIZE_MAX];
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@ -228,18 +229,6 @@ typedef struct {
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} RateModel;
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typedef struct {
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unsigned int SpaceAlloced;
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unsigned int MaxPermAlloced;
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double Tmp0[MAXFFTSIZE];
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double Tmp1[MAXFFTSIZE];
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double Tmp2[MAXFFTSIZE];
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double Tmp3[MAXFFTSIZE];
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int Perm[MAXFFTSIZE];
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int factor[NFACTOR];
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} FFTstr;
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/* The following strutc is used to store data from encoding, to make it
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fast and easy to construct a new bitstream with a different Bandwidth
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estimate. All values (except framelength and minBytes) is double size to
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