alacenc: pretty-printing and other cosmetics
This commit is contained in:
Justin Ruggles
parent
51c2483862
commit
fc9cf0b2a6
@ -119,12 +119,12 @@ static void encode_scalar(AlacEncodeContext *s, int x,
|
||||
|
||||
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
|
||||
{
|
||||
put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
|
||||
put_bits(&s->pbctx, 16, 0); // Seems to be zero
|
||||
put_bits(&s->pbctx, 1, 1); // Sample count is in the header
|
||||
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
|
||||
put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
|
||||
put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
|
||||
put_bits(&s->pbctx, 3, s->avctx->channels-1); // No. of channels -1
|
||||
put_bits(&s->pbctx, 16, 0); // Seems to be zero
|
||||
put_bits(&s->pbctx, 1, 1); // Sample count is in the header
|
||||
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
|
||||
put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim
|
||||
put_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
|
||||
}
|
||||
|
||||
static void calc_predictor_params(AlacEncodeContext *s, int ch)
|
||||
@ -167,8 +167,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
|
||||
/* calculate sum of 2nd order residual for each channel */
|
||||
sum[0] = sum[1] = sum[2] = sum[3] = 0;
|
||||
for (i = 2; i < n; i++) {
|
||||
lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
|
||||
rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
|
||||
lt = left_ch[i] - 2 * left_ch[i - 1] + left_ch[i - 2];
|
||||
rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];
|
||||
sum[2] += FFABS((lt + rt) >> 1);
|
||||
sum[3] += FFABS(lt - rt);
|
||||
sum[0] += FFABS(lt);
|
||||
@ -184,9 +184,8 @@ static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
|
||||
/* return mode with lowest score */
|
||||
best = 0;
|
||||
for (i = 1; i < 4; i++) {
|
||||
if (score[i] < score[best]) {
|
||||
if (score[i] < score[best])
|
||||
best = i;
|
||||
}
|
||||
}
|
||||
return best;
|
||||
}
|
||||
@ -199,40 +198,35 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s)
|
||||
|
||||
mode = estimate_stereo_mode(left, right, n);
|
||||
|
||||
switch(mode)
|
||||
{
|
||||
case ALAC_CHMODE_LEFT_RIGHT:
|
||||
s->interlacing_leftweight = 0;
|
||||
s->interlacing_shift = 0;
|
||||
break;
|
||||
|
||||
case ALAC_CHMODE_LEFT_SIDE:
|
||||
for (i = 0; i < n; i++) {
|
||||
right[i] = left[i] - right[i];
|
||||
}
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 0;
|
||||
break;
|
||||
|
||||
case ALAC_CHMODE_RIGHT_SIDE:
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = right[i];
|
||||
right[i] = left[i] - right[i];
|
||||
left[i] = tmp + (right[i] >> 31);
|
||||
}
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 31;
|
||||
break;
|
||||
|
||||
default:
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = left[i];
|
||||
left[i] = (tmp + right[i]) >> 1;
|
||||
right[i] = tmp - right[i];
|
||||
}
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 1;
|
||||
break;
|
||||
switch (mode) {
|
||||
case ALAC_CHMODE_LEFT_RIGHT:
|
||||
s->interlacing_leftweight = 0;
|
||||
s->interlacing_shift = 0;
|
||||
break;
|
||||
case ALAC_CHMODE_LEFT_SIDE:
|
||||
for (i = 0; i < n; i++)
|
||||
right[i] = left[i] - right[i];
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 0;
|
||||
break;
|
||||
case ALAC_CHMODE_RIGHT_SIDE:
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = right[i];
|
||||
right[i] = left[i] - right[i];
|
||||
left[i] = tmp + (right[i] >> 31);
|
||||
}
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 31;
|
||||
break;
|
||||
default:
|
||||
for (i = 0; i < n; i++) {
|
||||
tmp = left[i];
|
||||
left[i] = (tmp + right[i]) >> 1;
|
||||
right[i] = tmp - right[i];
|
||||
}
|
||||
s->interlacing_leftweight = 1;
|
||||
s->interlacing_shift = 1;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
@ -244,8 +238,10 @@ static void alac_linear_predictor(AlacEncodeContext *s, int 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];
|
||||
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;
|
||||
}
|
||||
@ -267,7 +263,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
|
||||
|
||||
for (j = 0; j < lpc.lpc_order; j++) {
|
||||
sum += (samples[lpc.lpc_order-j] - samples[0]) *
|
||||
lpc.lpc_coeff[j];
|
||||
lpc.lpc_coeff[j];
|
||||
}
|
||||
|
||||
sum >>= lpc.lpc_quant;
|
||||
@ -276,21 +272,20 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
|
||||
s->write_sample_size);
|
||||
res_val = residual[i];
|
||||
|
||||
if(res_val) {
|
||||
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];
|
||||
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;
|
||||
if (neg)
|
||||
sign *= -1;
|
||||
|
||||
lpc.lpc_coeff[index] -= sign;
|
||||
val *= sign;
|
||||
res_val -= ((val >> lpc.lpc_quant) *
|
||||
(lpc.lpc_order - index));
|
||||
res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);
|
||||
index--;
|
||||
}
|
||||
}
|
||||
@ -310,16 +305,16 @@ static void alac_entropy_coder(AlacEncodeContext *s)
|
||||
|
||||
k = av_log2((history >> 9) + 3);
|
||||
|
||||
x = -2*(*samples)-1;
|
||||
x ^= (x>>31);
|
||||
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);
|
||||
history += x * s->rc.history_mult -
|
||||
((history * s->rc.history_mult) >> 9);
|
||||
|
||||
sign_modifier = 0;
|
||||
if (x > 0xFFFF)
|
||||
@ -336,9 +331,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
|
||||
block_size++;
|
||||
}
|
||||
encode_scalar(s, block_size, k, 16);
|
||||
|
||||
sign_modifier = (block_size <= 0xFFFF);
|
||||
|
||||
history = 0;
|
||||
}
|
||||
|
||||
@ -356,7 +349,6 @@ static void write_compressed_frame(AlacEncodeContext *s)
|
||||
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
|
||||
|
||||
for (i = 0; i < s->avctx->channels; i++) {
|
||||
|
||||
calc_predictor_params(s, i);
|
||||
|
||||
put_bits(&s->pbctx, 4, prediction_type);
|
||||
@ -365,9 +357,8 @@ static void write_compressed_frame(AlacEncodeContext *s)
|
||||
put_bits(&s->pbctx, 3, s->rc.rice_modifier);
|
||||
put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
|
||||
// predictor coeff. table
|
||||
for (j = 0; j < s->lpc[i].lpc_order; j++) {
|
||||
for (j = 0; j < s->lpc[i].lpc_order; j++)
|
||||
put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
|
||||
}
|
||||
}
|
||||
|
||||
// apply lpc and entropy coding to audio samples
|
||||
@ -398,11 +389,11 @@ static av_cold int alac_encode_close(AVCodecContext *avctx)
|
||||
|
||||
static av_cold int alac_encode_init(AVCodecContext *avctx)
|
||||
{
|
||||
AlacEncodeContext *s = avctx->priv_data;
|
||||
AlacEncodeContext *s = avctx->priv_data;
|
||||
int ret;
|
||||
uint8_t *alac_extradata;
|
||||
|
||||
avctx->frame_size = DEFAULT_FRAME_SIZE;
|
||||
avctx->frame_size = DEFAULT_FRAME_SIZE;
|
||||
|
||||
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
|
||||
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
|
||||
@ -429,9 +420,11 @@ static av_cold int alac_encode_init(AVCodecContext *avctx)
|
||||
s->rc.k_modifier = 14;
|
||||
s->rc.rice_modifier = 4;
|
||||
|
||||
s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels * DEFAULT_SAMPLE_SIZE >> 3);
|
||||
s->max_coded_frame_size = 8 + (avctx->frame_size * avctx->channels *
|
||||
DEFAULT_SAMPLE_SIZE >> 3);
|
||||
|
||||
s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1; // FIXME: consider wasted_bytes
|
||||
// FIXME: consider wasted_bytes
|
||||
s->write_sample_size = DEFAULT_SAMPLE_SIZE + avctx->channels - 1;
|
||||
|
||||
avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
|
||||
if (!avctx->extradata) {
|
||||
@ -566,8 +559,8 @@ AVCodec ff_alac_encoder = {
|
||||
.init = alac_encode_init,
|
||||
.encode = alac_encode_frame,
|
||||
.close = alac_encode_close,
|
||||
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
|
||||
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
|
||||
AV_SAMPLE_FMT_NONE },
|
||||
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
||||
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
|
||||
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
|
||||
AV_SAMPLE_FMT_NONE },
|
||||
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
|
||||
};
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user