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alacenc: pretty-printing and other cosmetics

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This commit is contained in:
Justin Ruggles 2012-02-01 21:21:24 -05:00
parent 51c2483862
commit fc9cf0b2a6
1 changed files with 64 additions and 71 deletions
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135
libavcodec/alacenc.c
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@ -119,12 +119,12 @@ static void encode_scalar(AlacEncodeContext *s, int x,
static void write_frame_header(AlacEncodeContext *s, int is_verbatim) 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, 3, s->avctx->channels-1); // No. of channels -1
put_bits(&s->pbctx, 16, 0); // Seems to be zero 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, 1, 1); // Sample count is in the header
put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field put_bits(&s->pbctx, 2, 0); // FIXME: Wasted bytes field
put_bits(&s->pbctx, 1, is_verbatim); // Audio block is verbatim 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_bits32(&s->pbctx, s->avctx->frame_size); // No. of samples in the frame
} }
static void calc_predictor_params(AlacEncodeContext *s, int ch) 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 */ /* calculate sum of 2nd order residual for each channel */
sum[0] = sum[1] = sum[2] = sum[3] = 0; sum[0] = sum[1] = sum[2] = sum[3] = 0;
for (i = 2; i < n; i++) { for (i = 2; i < n; i++) {
lt = left_ch[i] - 2*left_ch[i-1] + left_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]; rt = right_ch[i] - 2 * right_ch[i - 1] + right_ch[i - 2];
sum[2] += FFABS((lt + rt) >> 1); sum[2] += FFABS((lt + rt) >> 1);
sum[3] += FFABS(lt - rt); sum[3] += FFABS(lt - rt);
sum[0] += FFABS(lt); 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 */ /* return mode with lowest score */
best = 0; best = 0;
for (i = 1; i < 4; i++) { for (i = 1; i < 4; i++) {
if (score[i] < score[best]) { if (score[i] < score[best])
best = i; best = i;
}
} }
return best; return best;
} }
@ -199,40 +198,35 @@ static void alac_stereo_decorrelation(AlacEncodeContext *s)
mode = estimate_stereo_mode(left, right, n); mode = estimate_stereo_mode(left, right, n);
switch(mode) switch (mode) {
{ case ALAC_CHMODE_LEFT_RIGHT:
case ALAC_CHMODE_LEFT_RIGHT: s->interlacing_leftweight = 0;
s->interlacing_leftweight = 0; s->interlacing_shift = 0;
s->interlacing_shift = 0; break;
break; case ALAC_CHMODE_LEFT_SIDE:
for (i = 0; i < n; i++)
case ALAC_CHMODE_LEFT_SIDE: right[i] = left[i] - right[i];
for (i = 0; i < n; i++) { s->interlacing_leftweight = 1;
right[i] = left[i] - right[i]; s->interlacing_shift = 0;
} break;
s->interlacing_leftweight = 1; case ALAC_CHMODE_RIGHT_SIDE:
s->interlacing_shift = 0; for (i = 0; i < n; i++) {
break; tmp = right[i];
right[i] = left[i] - right[i];
case ALAC_CHMODE_RIGHT_SIDE: left[i] = tmp + (right[i] >> 31);
for (i = 0; i < n; i++) { }
tmp = right[i]; s->interlacing_leftweight = 1;
right[i] = left[i] - right[i]; s->interlacing_shift = 31;
left[i] = tmp + (right[i] >> 31); break;
} default:
s->interlacing_leftweight = 1; for (i = 0; i < n; i++) {
s->interlacing_shift = 31; tmp = left[i];
break; left[i] = (tmp + right[i]) >> 1;
right[i] = tmp - right[i];
default: }
for (i = 0; i < n; i++) { s->interlacing_leftweight = 1;
tmp = left[i]; s->interlacing_shift = 1;
left[i] = (tmp + right[i]) >> 1; break;
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) { if (lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0]; s->predictor_buf[0] = s->sample_buf[ch][0];
for (i = 1; i < s->avctx->frame_size; i++) for (i = 1; i < s->avctx->frame_size; i++) {
s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1]; s->predictor_buf[i] = s->sample_buf[ch][i ] -
s->sample_buf[ch][i - 1];
}
return; return;
} }
@ -267,7 +263,7 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
for (j = 0; j < lpc.lpc_order; j++) { for (j = 0; j < lpc.lpc_order; j++) {
sum += (samples[lpc.lpc_order-j] - samples[0]) * sum += (samples[lpc.lpc_order-j] - samples[0]) *
lpc.lpc_coeff[j]; lpc.lpc_coeff[j];
} }
sum >>= lpc.lpc_quant; sum >>= lpc.lpc_quant;
@ -276,21 +272,20 @@ static void alac_linear_predictor(AlacEncodeContext *s, int ch)
s->write_sample_size); s->write_sample_size);
res_val = residual[i]; res_val = residual[i];
if(res_val) { if (res_val) {
int index = lpc.lpc_order - 1; int index = lpc.lpc_order - 1;
int neg = (res_val < 0); int neg = (res_val < 0);
while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) { while (index >= 0 && (neg ? (res_val < 0) : (res_val > 0))) {
int val = samples[0] - samples[lpc.lpc_order - index]; int val = samples[0] - samples[lpc.lpc_order - index];
int sign = (val ? FFSIGN(val) : 0); int sign = (val ? FFSIGN(val) : 0);
if(neg) if (neg)
sign*=-1; sign *= -1;
lpc.lpc_coeff[index] -= sign; lpc.lpc_coeff[index] -= sign;
val *= sign; val *= sign;
res_val -= ((val >> lpc.lpc_quant) * res_val -= (val >> lpc.lpc_quant) * (lpc.lpc_order - index);
(lpc.lpc_order - index));
index--; index--;
} }
} }
@ -310,16 +305,16 @@ static void alac_entropy_coder(AlacEncodeContext *s)
k = av_log2((history >> 9) + 3); k = av_log2((history >> 9) + 3);
x = -2*(*samples)-1; x = -2 * (*samples) -1;
x ^= (x>>31); x ^= x >> 31;
samples++; samples++;
i++; i++;
encode_scalar(s, x - sign_modifier, k, s->write_sample_size); encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
history += x * s->rc.history_mult history += x * s->rc.history_mult -
- ((history * s->rc.history_mult) >> 9); ((history * s->rc.history_mult) >> 9);
sign_modifier = 0; sign_modifier = 0;
if (x > 0xFFFF) if (x > 0xFFFF)
@ -336,9 +331,7 @@ static void alac_entropy_coder(AlacEncodeContext *s)
block_size++; block_size++;
} }
encode_scalar(s, block_size, k, 16); encode_scalar(s, block_size, k, 16);
sign_modifier = (block_size <= 0xFFFF); sign_modifier = (block_size <= 0xFFFF);
history = 0; history = 0;
} }
@ -356,7 +349,6 @@ static void write_compressed_frame(AlacEncodeContext *s)
put_bits(&s->pbctx, 8, s->interlacing_leftweight); put_bits(&s->pbctx, 8, s->interlacing_leftweight);
for (i = 0; i < s->avctx->channels; i++) { for (i = 0; i < s->avctx->channels; i++) {
calc_predictor_params(s, i); calc_predictor_params(s, i);
put_bits(&s->pbctx, 4, prediction_type); 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, 3, s->rc.rice_modifier);
put_bits(&s->pbctx, 5, s->lpc[i].lpc_order); put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
// predictor coeff. table // 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]); put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
}
} }
// apply lpc and entropy coding to audio samples // 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) static av_cold int alac_encode_init(AVCodecContext *avctx)
{ {
AlacEncodeContext *s = avctx->priv_data; AlacEncodeContext *s = avctx->priv_data;
int ret; int ret;
uint8_t *alac_extradata; uint8_t *alac_extradata;
avctx->frame_size = DEFAULT_FRAME_SIZE; avctx->frame_size = DEFAULT_FRAME_SIZE;
if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n"); 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.k_modifier = 14;
s->rc.rice_modifier = 4; 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); avctx->extradata = av_mallocz(ALAC_EXTRADATA_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
if (!avctx->extradata) { if (!avctx->extradata) {
@ -566,8 +559,8 @@ AVCodec ff_alac_encoder = {
.init = alac_encode_init, .init = alac_encode_init,
.encode = alac_encode_frame, .encode = alac_encode_frame,
.close = alac_encode_close, .close = alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME, .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
}; };