* ALAC audio encoder
* Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#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
int rice_modifier;
} RiceContext;
-typedef struct LPCContext {
+typedef struct AlacLPCContext {
int lpc_order;
int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
int lpc_quant;
-} LPCContext;
+} AlacLPCContext;
typedef struct AlacEncodeContext {
int compression_level;
int interlacing_leftweight;
PutBitContext pbctx;
RiceContext rc;
- LPCContext lpc[MAX_CHANNELS];
- DSPContext dspctx;
+ AlacLPCContext lpc[MAX_CHANNELS];
+ LPCContext lpc_ctx;
AVCodecContext *avctx;
} 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, DEFAULT_MIN_PRED_ORDER, DEFAULT_MAX_PRED_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->lpc_ctx, s->sample_buf[ch],
+ s->avctx->frame_size,
+ s->min_prediction_order,
+ s->max_prediction_order,
+ ALAC_MAX_LPC_PRECISION, coefs, shift,
+ FF_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)
}
}
+static void alac_linear_predictor(AlacEncodeContext *s, int ch)
+{
+ int i;
+ AlacLPCContext 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] = sign_extend(samples[lpc.lpc_order+1] - sum,
+ s->write_sample_size);
+ 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;
- /* only simple mid/side decorrelation supported as of now */
- alac_stereo_decorrelation(s);
+ if(s->avctx->channels == 2)
+ alac_stereo_decorrelation(s);
put_bits(&s->pbctx, 8, s->interlacing_shift);
put_bits(&s->pbctx, 8, s->interlacing_leftweight);
static av_cold int alac_encode_init(AVCodecContext *avctx)
{
AlacEncodeContext *s = avctx->priv_data;
+ int ret;
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) {
+ if(avctx->sample_fmt != AV_SAMPLE_FMT_S16) {
av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
return -1;
}
// 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
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) {
+ avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
return -1;
}
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) {
+ avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
return -1;
}
avctx->coded_frame->key_frame = 1;
s->avctx = avctx;
- dsputil_init(&s->dspctx, avctx);
+ ret = ff_lpc_init(&s->lpc_ctx, avctx->frame_size, s->max_prediction_order,
+ FF_LPC_TYPE_LEVINSON);
- return 0;
+ return ret;
}
static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
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++);
static av_cold int alac_encode_close(AVCodecContext *avctx)
{
+ AlacEncodeContext *s = avctx->priv_data;
+ ff_lpc_end(&s->lpc_ctx);
av_freep(&avctx->extradata);
avctx->extradata_size = 0;
av_freep(&avctx->coded_frame);
return 0;
}
-AVCodec alac_encoder = {
+AVCodec ff_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 AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff