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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

misc spelling/wording/grammar fixes

Originally committed as revision 15296 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Diego Biurrun 2008-09-11 08:13:23 +00:00
parent 77298e99b7
commit 36b3e36e00
3 changed files with 26 additions and 26 deletions

View File

@ -38,16 +38,16 @@ typedef struct {
unsigned int lpc_tables[2][10]; unsigned int lpc_tables[2][10];
/** LPC coefficients: lpc_coef[0] is the coefficients of the current frame /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame
* and lpc_coef[1] of the previous one */ * and lpc_coef[1] of the previous one. */
unsigned int *lpc_coef[2]; unsigned int *lpc_coef[2];
unsigned int lpc_refl_rms[2]; unsigned int lpc_refl_rms[2];
/** the current subblock padded by the last 10 values of the previous one*/ /** The current subblock padded by the last 10 values of the previous one. */
int16_t curr_sblock[50]; int16_t curr_sblock[50];
/** adaptive codebook. Its size is two units bigger to avoid a /** Adaptive codebook, its size is two units bigger to avoid a
* buffer overflow */ * buffer overflow. */
uint16_t adapt_cb[146+2]; uint16_t adapt_cb[146+2];
} RA144Context; } RA144Context;
@ -218,7 +218,7 @@ static void int_to_int16(int16_t *out, const int *inp)
* Evaluate the reflection coefficients from the filter coefficients. * Evaluate the reflection coefficients from the filter coefficients.
* Does the inverse of the eval_coefs() function. * Does the inverse of the eval_coefs() function.
* *
* @return 1 if one of the reflection coefficients is of magnitude greater than * @return 1 if one of the reflection coefficients is greater than
* 4095, 0 if not. * 4095, 0 if not.
*/ */
static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx) static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx)
@ -265,14 +265,14 @@ static int interp(RA144Context *ractx, int16_t *out, int a,
int b = NBLOCKS - a; int b = NBLOCKS - a;
int i; int i;
// Interpolate block coefficients from the this frame forth block and // Interpolate block coefficients from the this frame's forth block and
// last frame forth block // last frame's forth block.
for (i=0; i<30; i++) for (i=0; i<30; i++)
out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2; out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2;
if (eval_refl(work, out, ractx)) { if (eval_refl(work, out, ractx)) {
// The interpolated coefficients are unstable, copy either new or old // The interpolated coefficients are unstable, copy either new or old
// coefficients // coefficients.
int_to_int16(out, ractx->lpc_coef[copyold]); int_to_int16(out, ractx->lpc_coef[copyold]);
return rescale_rms(ractx->lpc_refl_rms[copyold], energy); return rescale_rms(ractx->lpc_refl_rms[copyold], energy);
} else { } else {
@ -280,7 +280,7 @@ static int interp(RA144Context *ractx, int16_t *out, int a,
} }
} }
/** Uncompress one block (20 bytes -> 160*2 bytes) */ /** Uncompress one block (20 bytes -> 160*2 bytes). */
static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, static int ra144_decode_frame(AVCodecContext * avctx, void *vdata,
int *data_size, const uint8_t *buf, int buf_size) int *data_size, const uint8_t *buf, int buf_size)
{ {

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@ -29,18 +29,18 @@ typedef struct {
float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
float sp_hist[111]; ///< Speech data history (spec: SB) float sp_hist[111]; ///< speech data history (spec: SB)
/** Speech part of the gain autocorrelation (spec: REXP) */ /** speech part of the gain autocorrelation (spec: REXP) */
float sp_rec[37]; float sp_rec[37];
float gain_hist[38]; ///< Log-gain history (spec: SBLG) float gain_hist[38]; ///< log-gain history (spec: SBLG)
/** Recursive part of the gain autocorrelation (spec: REXPLG) */ /** recursive part of the gain autocorrelation (spec: REXPLG) */
float gain_rec[11]; float gain_rec[11];
float sp_block[41]; ///< Speech data of four blocks (spec: STTMP) float sp_block[41]; ///< four blocks of speech data (spec: STTMP)
float gain_block[10]; ///< Gain data of four blocks (spec: GSTATE) float gain_block[10]; ///< four blocks of gain data (spec: GSTATE)
} RA288Context; } RA288Context;
static av_cold int ra288_decode_init(AVCodecContext *avctx) static av_cold int ra288_decode_init(AVCodecContext *avctx)
@ -71,7 +71,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
int i, j; int i, j;
double sumsum; double sumsum;
float sum, buffer[5]; float sum, buffer[5];
float *block = ractx->sp_block + 36; // Current block float *block = ractx->sp_block + 36; // current block
memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block)); memmove(ractx->sp_block, ractx->sp_block + 5, 36*sizeof(*ractx->sp_block));
@ -122,14 +122,14 @@ static void convolve(float *tgt, const float *src, int len, int n)
} }
/** /**
* Hybrid window filtering. See blocks 36 and 49 of the G.728 specification. * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
* *
* @param order the order of the filter * @param order filter order
* @param n the length of the input * @param n input length
* @param non_rec the number of non-recursive samples * @param non_rec number of non-recursive samples
* @param out the filter output * @param out filter output
* @param in pointer to the input of the filter * @param in pointer to the input of the filter
* @param hist pointer to the input history of the filter. It is updated by * @param hist Pointer to the input history of the filter, it is updated by
* this function. * this function.
* @param out pointer to the non-recursive part of the output * @param out pointer to the non-recursive part of the output
* @param out2 pointer to the recursive part of the output * @param out2 pointer to the recursive part of the output
@ -158,12 +158,12 @@ static void do_hybrid_window(int order, int n, int non_rec, const float *in,
out [i] = out2[i] + buffer2[i]; out [i] = out2[i] + buffer2[i];
} }
/* Multiply by the white noise correcting factor (WNCF) */ /* Multiply by the white noise correcting factor (WNCF). */
*out *= 257./256.; *out *= 257./256.;
} }
/** /**
* Backward synthesis filter. Find the LPC coefficients from past speech data. * Backward synthesis filter, find the LPC coefficients from past speech data.
*/ */
static void backward_filter(RA288Context *ractx) static void backward_filter(RA288Context *ractx)
{ {

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@ -126,7 +126,7 @@ static const float gain_window[38]={
0.183868408, 0.0923461914 0.183868408, 0.0923461914
}; };
/** Synthesis bandwidth broadening table */ /** synthesis bandwidth broadening table */
static const float syn_bw_tab[36]={ static const float syn_bw_tab[36]={
0.98828125, 0.976699829, 0.965254128, 0.953942537, 0.942763507, 0.931715488, 0.98828125, 0.976699829, 0.965254128, 0.953942537, 0.942763507, 0.931715488,
0.920796931, 0.910006344, 0.899342179, 0.888803005, 0.878387332, 0.868093729, 0.920796931, 0.910006344, 0.899342179, 0.888803005, 0.878387332, 0.868093729,
@ -136,7 +136,7 @@ static const float syn_bw_tab[36]={
0.693900526, 0.685768902, 0.677732527, 0.669790328, 0.66194123, 0.654184103 0.693900526, 0.685768902, 0.677732527, 0.669790328, 0.66194123, 0.654184103
}; };
/** Gain bandwidth broadening table */ /** gain bandwidth broadening table */
static const float gain_bw_tab[10]={ static const float gain_bw_tab[10]={
0.90625, 0.821289063, 0.74432373, 0.674499512, 0.61126709, 0.90625, 0.821289063, 0.74432373, 0.674499512, 0.61126709,
0.553955078, 0.50201416, 0.454956055, 0.41229248, 0.373657227 0.553955078, 0.50201416, 0.454956055, 0.41229248, 0.373657227