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misc spelling/wording/grammar fixes
Originally committed as revision 15296 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -38,16 +38,16 @@ typedef struct {
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unsigned int lpc_tables[2][10];
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/** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
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* and lpc_coef[1] of the previous one */
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* and lpc_coef[1] of the previous one. */
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unsigned int *lpc_coef[2];
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unsigned int lpc_refl_rms[2];
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/** the current subblock padded by the last 10 values of the previous one*/
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/** The current subblock padded by the last 10 values of the previous one. */
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int16_t curr_sblock[50];
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/** adaptive codebook. Its size is two units bigger to avoid a
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* buffer overflow */
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/** Adaptive codebook, its size is two units bigger to avoid a
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* buffer overflow. */
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uint16_t adapt_cb[146+2];
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} RA144Context;
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@ -218,7 +218,7 @@ static void int_to_int16(int16_t *out, const int *inp)
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* Evaluate the reflection coefficients from the filter coefficients.
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* Does the inverse of the eval_coefs() function.
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*
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* @return 1 if one of the reflection coefficients is of magnitude greater than
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* @return 1 if one of the reflection coefficients is greater than
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* 4095, 0 if not.
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*/
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static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx)
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@ -265,14 +265,14 @@ static int interp(RA144Context *ractx, int16_t *out, int a,
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int b = NBLOCKS - a;
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int i;
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// Interpolate block coefficients from the this frame forth block and
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// last frame forth block
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// Interpolate block coefficients from the this frame's forth block and
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// last frame's forth block.
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for (i=0; i<30; i++)
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out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;
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if (eval_refl(work, out, ractx)) {
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// The interpolated coefficients are unstable, copy either new or old
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// coefficients
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// coefficients.
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int_to_int16(out, ractx->lpc_coef[copyold]);
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return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
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} else {
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@ -280,7 +280,7 @@ static int interp(RA144Context *ractx, int16_t *out, int a,
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}
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}
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/** Uncompress one block (20 bytes -> 160*2 bytes) */
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/** Uncompress one block (20 bytes -> 160*2 bytes). */
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static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
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int *data_size, const uint8_t *buf, int buf_size)
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{
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@ -29,18 +29,18 @@ typedef struct {
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float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
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float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
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float sp_hist[111]; ///< Speech data history (spec: SB)
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float sp_hist[111]; ///< speech data history (spec: SB)
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/** Speech part of the gain autocorrelation (spec: REXP) */
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/** speech part of the gain autocorrelation (spec: REXP) */
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float sp_rec[37];
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float gain_hist[38]; ///< Log-gain history (spec: SBLG)
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float gain_hist[38]; ///< log-gain history (spec: SBLG)
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/** Recursive part of the gain autocorrelation (spec: REXPLG) */
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/** recursive part of the gain autocorrelation (spec: REXPLG) */
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float gain_rec[11];
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float sp_block[41]; ///< Speech data of four blocks (spec: STTMP)
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float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE)
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float sp_block[41]; ///< four blocks of speech data (spec: STTMP)
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float gain_block[10]; ///< four blocks of gain data (spec: GSTATE)
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} RA288Context;
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static av_cold int ra288_decode_init(AVCodecContext *avctx)
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@ -71,7 +71,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
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int i, j;
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double sumsum;
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float sum, buffer[5];
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float *block = ractx->sp_block + 36; // Current block
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float *block = ractx->sp_block + 36; // current block
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memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block));
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@ -122,14 +122,14 @@ static void convolve(float *tgt, const float *src, int len, int n)
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}
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/**
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* Hybrid window filtering. See blocks 36 and 49 of the G.728 specification.
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* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
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*
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* @param order the order of the filter
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* @param n the length of the input
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* @param non_rec the number of non-recursive samples
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* @param out the filter output
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* @param order filter order
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* @param n input length
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* @param non_rec number of non-recursive samples
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* @param out filter output
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* @param in pointer to the input of the filter
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* @param hist pointer to the input history of the filter. It is updated by
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* @param hist Pointer to the input history of the filter, it is updated by
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* this function.
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* @param out pointer to the non-recursive part of the output
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* @param out2 pointer to the recursive part of the output
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@ -158,12 +158,12 @@ static void do_hybrid_window(int order, int n, int non_rec, const float *in,
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out [i] = out2[i] + buffer2[i];
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}
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/* Multiply by the white noise correcting factor (WNCF) */
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/* Multiply by the white noise correcting factor (WNCF). */
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*out *= 257./256.;
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}
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/**
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* Backward synthesis filter. Find the LPC coefficients from past speech data.
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* Backward synthesis filter, find the LPC coefficients from past speech data.
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*/
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static void backward_filter(RA288Context *ractx)
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{
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@ -126,7 +126,7 @@ static const float gain_window[38]={
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0.183868408, 0.0923461914
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};
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/** Synthesis bandwidth broadening table */
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/** synthesis bandwidth broadening table */
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static const float syn_bw_tab[36]={
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0.98828125, 0.976699829, 0.965254128, 0.953942537, 0.942763507, 0.931715488,
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0.920796931, 0.910006344, 0.899342179, 0.888803005, 0.878387332, 0.868093729,
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@ -136,7 +136,7 @@ static const float syn_bw_tab[36]={
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0.693900526, 0.685768902, 0.677732527, 0.669790328, 0.66194123, 0.654184103
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};
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/** Gain bandwidth broadening table */
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/** gain bandwidth broadening table */
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static const float gain_bw_tab[10]={
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0.90625, 0.821289063, 0.74432373, 0.674499512, 0.61126709,
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0.553955078, 0.50201416, 0.454956055, 0.41229248, 0.373657227
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