* 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"
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, 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->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;
- LPCContext lpc = s->lpc[ch];
+ AlacLPCContext lpc = s->lpc[ch];
if(lpc.lpc_order == 31) {
s->predictor_buf[0] = s->sample_buf[ch][0];
{
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);
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_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_coded_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_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
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 = {
- "alac",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ALAC,
- sizeof(AlacEncodeContext),
- alac_encode_init,
- alac_encode_frame,
- alac_encode_close,
+AVCodec ff_alac_encoder = {
+ .name = "alac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_ALAC,
+ .priv_data_size = sizeof(AlacEncodeContext),
+ .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)"),
};
projects / ffmpeg / blobdiff