*/
#include "avcodec.h"
-#include "bitstream.h"
+#include "put_bits.h"
#include "dsputil.h"
#include "lpc.h"
+#include "mathops.h"
#define DEFAULT_FRAME_SIZE 4096
#define DEFAULT_SAMPLE_SIZE 16
} AlacEncodeContext;
-static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
+static void init_sample_buffers(AlacEncodeContext *s, const int16_t *input_samples)
{
int ch, i;
for(ch=0;ch<s->avctx->channels;ch++) {
- int16_t *sptr = input_samples + ch;
+ const int16_t *sptr = input_samples + ch;
for(i=0;i<s->avctx->frame_size;i++) {
s->sample_buf[ch][i] = *sptr;
sptr += s->avctx->channels;
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_bits(&s->pbctx, 32, 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)
int shift[MAX_LPC_ORDER];
int opt_order;
- opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch], s->avctx->frame_size, s->min_prediction_order, s->max_prediction_order,
- ALAC_MAX_LPC_PRECISION, coefs, shift, 1, ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
-
- s->lpc[ch].lpc_order = opt_order;
- s->lpc[ch].lpc_quant = shift[opt_order-1];
- memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
+ if (s->compression_level == 1) {
+ s->lpc[ch].lpc_order = 6;
+ s->lpc[ch].lpc_quant = 6;
+ s->lpc[ch].lpc_coeff[0] = 160;
+ s->lpc[ch].lpc_coeff[1] = -190;
+ s->lpc[ch].lpc_coeff[2] = 170;
+ s->lpc[ch].lpc_coeff[3] = -130;
+ s->lpc[ch].lpc_coeff[4] = 80;
+ s->lpc[ch].lpc_coeff[5] = -25;
+ } else {
+ opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
+ s->avctx->frame_size,
+ s->min_prediction_order,
+ s->max_prediction_order,
+ ALAC_MAX_LPC_PRECISION, coefs, shift,
+ AV_LPC_TYPE_LEVINSON, 0,
+ ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
+
+ s->lpc[ch].lpc_order = opt_order;
+ s->lpc[ch].lpc_quant = shift[opt_order-1];
+ memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
+ }
}
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
sum >>= lpc.lpc_quant;
sum += samples[0];
- residual[i] = samples[lpc.lpc_order+1] - sum;
+ residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
+ s->write_sample_size);
res_val = residual[i];
if(res_val) {
{
int i, j;
- /* only simple mid/side decorrelation supported as of now */
if(s->avctx->channels == 2)
alac_stereo_decorrelation(s);
put_bits(&s->pbctx, 8, s->interlacing_shift);
uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
avctx->frame_size = DEFAULT_FRAME_SIZE;
- avctx->bits_per_sample = DEFAULT_SAMPLE_SIZE;
+ avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
if(avctx->sample_fmt != SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
// Set default compression level
if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
- s->compression_level = 1;
+ s->compression_level = 2;
else
- s->compression_level = av_clip(avctx->compression_level, 0, 1);
+ s->compression_level = av_clip(avctx->compression_level, 0, 2);
// Initialize default Rice parameters
s->rc.history_mult = 40;
s->rc.k_modifier = 14;
s->rc.rice_modifier = 4;
- s->max_coded_frame_size = (ALAC_FRAME_HEADER_SIZE + ALAC_FRAME_FOOTER_SIZE +
- avctx->frame_size*avctx->channels*avctx->bits_per_sample)>>3;
+ s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
- s->write_sample_size = avctx->bits_per_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
+ s->write_sample_size = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
AV_WB32(alac_extradata, ALAC_EXTRADATA_SIZE);
AV_WB32(alac_extradata+4, MKBETAG('a','l','a','c'));
AV_WB32(alac_extradata+12, avctx->frame_size);
- AV_WB8 (alac_extradata+17, avctx->bits_per_sample);
+ AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
AV_WB8 (alac_extradata+21, avctx->channels);
AV_WB32(alac_extradata+24, s->max_coded_frame_size);
- AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_sample); // average bitrate
+ AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
AV_WB32(alac_extradata+32, avctx->sample_rate);
// Set relevant extradata fields
if((s->compression_level == 0) || verbatim_flag) {
// Verbatim mode
- int16_t *samples = data;
+ const int16_t *samples = data;
write_frame_header(s, 1);
for(i=0; i<avctx->frame_size*avctx->channels; i++) {
put_sbits(pb, 16, *samples++);
AVCodec alac_encoder = {
"alac",
- CODEC_TYPE_AUDIO,
+ AVMEDIA_TYPE_AUDIO,
CODEC_ID_ALAC,
sizeof(AlacEncodeContext),
alac_encode_init,
alac_encode_frame,
alac_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
+ .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};