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alacenc: last few hunks approved by michael

Originally committed as revision 14845 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Jai Menon 2008-08-19 17:20:41 +00:00
parent cc940caf4d
commit e13894e80e
3 changed files with 112 additions and 1 deletions

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@ -29,6 +29,7 @@ OBJS-$(CONFIG_AASC_DECODER) += aasc.o
OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3tab.o ac3dec_data.o ac3.o mdct.o fft.o
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o lpc.o
OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_ASV1_DECODER) += asv1.o mpeg12data.o

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@ -217,6 +217,114 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s)
}
}
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
{
int i;
LPCContext lpc = s->lpc[ch];
if(lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
for(i=1; i<s->avctx->frame_size; i++)
s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
return;
}
// generalised linear predictor
if(lpc.lpc_order > 0) {
int32_t *samples = s->sample_buf[ch];
int32_t *residual = s->predictor_buf;
// generate warm-up samples
residual[0] = samples[0];
for(i=1;i<=lpc.lpc_order;i++)
residual[i] = samples[i] - samples[i-1];
// perform lpc on remaining samples
for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
for (j = 0; j < lpc.lpc_order; j++) {
sum += (samples[lpc.lpc_order-j] - samples[0]) *
lpc.lpc_coeff[j];
}
sum >>= lpc.lpc_quant;
sum += samples[0];
residual[i] = samples[lpc.lpc_order+1] - sum;
res_val = residual[i];
if(res_val) {
int index = lpc.lpc_order - 1;
int neg = (res_val < 0);
while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
int val = samples[0] - samples[lpc.lpc_order - index];
int sign = (val ? FFSIGN(val) : 0);
if(neg)
sign*=-1;
lpc.lpc_coeff[index] -= sign;
val *= sign;
res_val -= ((val >> lpc.lpc_quant) *
(lpc.lpc_order - index));
index--;
}
}
samples++;
}
}
}
static void alac_entropy_coder(AlacEncodeContext *s)
{
unsigned int history = s->rc.initial_history;
int sign_modifier = 0, i, k;
int32_t *samples = s->predictor_buf;
for(i=0;i < s->avctx->frame_size;) {
int x;
k = av_log2((history >> 9) + 3);
x = -2*(*samples)-1;
x ^= (x>>31);
samples++;
i++;
encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
history += x * s->rc.history_mult
- ((history * s->rc.history_mult) >> 9);
sign_modifier = 0;
if(x > 0xFFFF)
history = 0xFFFF;
if((history < 128) && (i < s->avctx->frame_size)) {
unsigned int block_size = 0;
k = 7 - av_log2(history) + ((history + 16) >> 6);
while((*samples == 0) && (i < s->avctx->frame_size)) {
samples++;
i++;
block_size++;
}
encode_scalar(s, block_size, k, 16);
sign_modifier = (block_size <= 0xFFFF);
history = 0;
}
}
}
static void write_compressed_frame(AlacEncodeContext *s)
{
int i, j;
@ -295,6 +403,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
AV_WB8(alac_extradata+20, s->rc.k_modifier);
}
s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
if(avctx->min_prediction_order >= 0) {
if(avctx->min_prediction_order < MIN_LPC_ORDER ||
avctx->min_prediction_order > MAX_LPC_ORDER) {
@ -305,6 +414,7 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
s->min_prediction_order = avctx->min_prediction_order;
}
s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
if(avctx->max_prediction_order >= 0) {
if(avctx->max_prediction_order < MIN_LPC_ORDER ||
avctx->max_prediction_order > MAX_LPC_ORDER) {

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@ -182,7 +182,7 @@ void avcodec_register_all(void)
/* audio codecs */
REGISTER_ENCDEC (AC3, ac3);
REGISTER_DECODER (ALAC, alac);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (APE, ape);
REGISTER_DECODER (ATRAC3, atrac3);
REGISTER_DECODER (COOK, cook);