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Merge codec_consistency branch. This should make sample usage much happier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -44,9 +44,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#define BUFFER_SAMPLES 8096 /* size for the translation buffers */
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/* Sample frame data */
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#include "slin_adpcm_ex.h"
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#include "adpcm_slin_ex.h"
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#include "asterisk/slin.h"
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#include "ex_adpcm.h"
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/*
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* Step size index shift table
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@@ -285,43 +284,12 @@ static struct ast_frame *lintoadpcm_frameout(struct ast_trans_pvt *pvt)
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}
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/*! \brief AdpcmToLin_Sample */
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static struct ast_frame *adpcmtolin_sample(void)
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{
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static struct ast_frame f;
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f.frametype = AST_FRAME_VOICE;
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f.subclass = AST_FORMAT_ADPCM;
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f.datalen = sizeof(adpcm_slin_ex);
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f.samples = sizeof(adpcm_slin_ex) * 2;
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f.mallocd = 0;
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f.offset = 0;
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f.src = __PRETTY_FUNCTION__;
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f.data.ptr = adpcm_slin_ex;
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return &f;
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}
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/*! \brief LinToAdpcm_Sample */
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static struct ast_frame *lintoadpcm_sample(void)
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{
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static struct ast_frame f;
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f.frametype = AST_FRAME_VOICE;
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f.subclass = AST_FORMAT_SLINEAR;
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f.datalen = sizeof(slin_adpcm_ex);
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/* Assume 8000 Hz */
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f.samples = sizeof(slin_adpcm_ex) / 2;
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f.mallocd = 0;
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f.offset = 0;
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f.src = __PRETTY_FUNCTION__;
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f.data.ptr = slin_adpcm_ex;
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return &f;
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}
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static struct ast_translator adpcmtolin = {
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.name = "adpcmtolin",
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.srcfmt = AST_FORMAT_ADPCM,
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.dstfmt = AST_FORMAT_SLINEAR,
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.framein = adpcmtolin_framein,
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.sample = adpcmtolin_sample,
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.sample = adpcm_sample,
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.desc_size = sizeof(struct adpcm_decoder_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES * 2,
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@@ -334,7 +302,7 @@ static struct ast_translator lintoadpcm = {
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.dstfmt = AST_FORMAT_ADPCM,
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.framein = lintoadpcm_framein,
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.frameout = lintoadpcm_frameout,
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.sample = lintoadpcm_sample,
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.sample = slin8_sample,
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.desc_size = sizeof (struct adpcm_encoder_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES/ 2, /* 2 samples per byte */
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