* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
*
- * 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
*/
/**
- * @file libavcodec/alac.c
+ * @file
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
- *
- * For more information on the ALAC format, visit:
- * http://crazney.net/programs/itunes/alac.html
+ * @see http://crazney.net/programs/itunes/alac.html
*
* Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
#include "mathops.h"
AVCodecContext *avctx;
GetBitContext gb;
- /* init to 0; first frame decode should initialize from extradata and
- * set this to 1 */
- int context_initialized;
int numchannels;
int bytespersample;
int32_t *outputsamples_buffer[MAX_CHANNELS];
+ int32_t *wasted_bits_buffer[MAX_CHANNELS];
+
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
uint8_t setinfo_sample_size; /* 0x10 */
uint8_t setinfo_rice_kmodifier; /* 0x0e */
/* end setinfo stuff */
+ int wasted_bits;
} ALACContext;
static void allocate_buffers(ALACContext *alac)
alac->outputsamples_buffer[chan] =
av_malloc(alac->setinfo_max_samples_per_frame * 4);
+
+ alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
}
}
buffer_out[i+1] = val;
}
-#if 0
/* 4 and 8 are very common cases (the only ones i've seen). these
* should be unrolled and optimized
*/
- if (predictor_coef_num == 4) {
- /* FIXME: optimized general case */
- return;
- }
-
- if (predictor_coef_table == 8) {
- /* FIXME: optimized general case */
- return;
- }
-#endif
/* general case */
if (predictor_coef_num > 0) {
}
}
+static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS],
+ int32_t *buffer_out,
+ int32_t *wasted_bits_buffer[MAX_CHANNELS],
+ int wasted_bits,
+ int numchannels, int numsamples,
+ uint8_t interlacing_shift,
+ uint8_t interlacing_leftweight)
+{
+ int i;
+
+ if (numsamples <= 0)
+ return;
+
+ /* weighted interlacing */
+ if (interlacing_leftweight) {
+ for (i = 0; i < numsamples; i++) {
+ int32_t a, b;
+
+ a = buffer[0][i];
+ b = buffer[1][i];
+
+ a -= (b * interlacing_leftweight) >> interlacing_shift;
+ b += a;
+
+ if (wasted_bits) {
+ b = (b << wasted_bits) | wasted_bits_buffer[0][i];
+ a = (a << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = b << 8;
+ buffer_out[i * numchannels + 1] = a << 8;
+ }
+ } else {
+ for (i = 0; i < numsamples; i++) {
+ int32_t left, right;
+
+ left = buffer[0][i];
+ right = buffer[1][i];
+
+ if (wasted_bits) {
+ left = (left << wasted_bits) | wasted_bits_buffer[0][i];
+ right = (right << wasted_bits) | wasted_bits_buffer[1][i];
+ }
+
+ buffer_out[i * numchannels] = left << 8;
+ buffer_out[i * numchannels + 1] = right << 8;
+ }
+ }
+}
+
static int alac_decode_frame(AVCodecContext *avctx,
void *outbuffer, int *outputsize,
- const uint8_t *inbuffer, int input_buffer_size)
+ AVPacket *avpkt)
{
+ const uint8_t *inbuffer = avpkt->data;
+ int input_buffer_size = avpkt->size;
ALACContext *alac = avctx->priv_data;
int channels;
unsigned int outputsamples;
int hassize;
unsigned int readsamplesize;
- int wasted_bytes;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
/* short-circuit null buffers */
if (!inbuffer || !input_buffer_size)
- return input_buffer_size;
-
- /* initialize from the extradata */
- if (!alac->context_initialized) {
- if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
- av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
- ALAC_EXTRADATA_SIZE);
- return input_buffer_size;
- }
- if (alac_set_info(alac)) {
- av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
- return input_buffer_size;
- }
- alac->context_initialized = 1;
- }
+ return -1;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
if (channels > MAX_CHANNELS) {
av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
MAX_CHANNELS);
- return input_buffer_size;
+ return -1;
}
/* 2^result = something to do with output waiting.
/* the output sample size is stored soon */
hassize = get_bits1(&alac->gb);
- wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
+ alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
} else
outputsamples = alac->setinfo_max_samples_per_frame;
+ switch (alac->setinfo_sample_size) {
+ case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ alac->bytespersample = channels << 1;
+ break;
+ case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ alac->bytespersample = channels << 2;
+ break;
+ default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n",
+ alac->setinfo_sample_size);
+ return -1;
+ }
+
if(outputsamples > *outputsize / alac->bytespersample){
av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
return -1;
}
*outputsize = outputsamples * alac->bytespersample;
- readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
+ readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
return -1;
if (!isnotcompressed) {
/* so it is compressed */
- int16_t predictor_coef_table[channels][32];
- int predictor_coef_num[channels];
- int prediction_type[channels];
- int prediction_quantitization[channels];
- int ricemodifier[channels];
+ int16_t predictor_coef_table[MAX_CHANNELS][32];
+ int predictor_coef_num[MAX_CHANNELS];
+ int prediction_type[MAX_CHANNELS];
+ int prediction_quantitization[MAX_CHANNELS];
+ int ricemodifier[MAX_CHANNELS];
int i, chan;
interlacing_shift = get_bits(&alac->gb, 8);
predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
}
- if (wasted_bytes)
- av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
-
+ if (alac->wasted_bits) {
+ int i, ch;
+ for (i = 0; i < outputsamples; i++) {
+ for (ch = 0; ch < channels; ch++)
+ alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits);
+ }
+ }
for (chan = 0; chan < channels; chan++) {
bastardized_rice_decompress(alac,
alac->predicterror_buffer[chan],
} else {
/* not compressed, easy case */
int i, chan;
+ if (alac->setinfo_sample_size <= 16) {
for (i = 0; i < outputsamples; i++)
for (chan = 0; chan < channels; chan++) {
int32_t audiobits;
alac->outputsamples_buffer[chan][i] = audiobits;
}
- /* wasted_bytes = 0; */
+ } else {
+ for (i = 0; i < outputsamples; i++) {
+ for (chan = 0; chan < channels; chan++) {
+ alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb,
+ alac->setinfo_sample_size);
+ alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i],
+ alac->setinfo_sample_size);
+ }
+ }
+ }
+ alac->wasted_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
} else {
int i;
for (i = 0; i < outputsamples; i++) {
- int16_t sample = alac->outputsamples_buffer[0][i];
- ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
+ ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
}
}
break;
- case 20:
case 24:
- // It is not clear if there exist any encoder that creates 24 bit ALAC
- // files. iTunes convert 24 bit raw files to 16 bit before encoding.
- case 32:
- av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
- break;
- default:
+ if (channels == 2) {
+ decorrelate_stereo_24(alac->outputsamples_buffer,
+ outbuffer,
+ alac->wasted_bits_buffer,
+ alac->wasted_bits,
+ alac->numchannels,
+ outputsamples,
+ interlacing_shift,
+ interlacing_leftweight);
+ } else {
+ int i;
+ for (i = 0; i < outputsamples; i++)
+ ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
+ }
break;
}
{
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
- alac->context_initialized = 0;
-
alac->numchannels = alac->avctx->channels;
- alac->bytespersample = 2 * alac->numchannels;
- avctx->sample_fmt = SAMPLE_FMT_S16;
+
+ /* initialize from the extradata */
+ if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
+ ALAC_EXTRADATA_SIZE);
+ return -1;
+ }
+ if (alac_set_info(alac)) {
+ av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
+ return -1;
+ }
return 0;
}
int chan;
for (chan = 0; chan < MAX_CHANNELS; chan++) {
- av_free(alac->predicterror_buffer[chan]);
- av_free(alac->outputsamples_buffer[chan]);
+ av_freep(&alac->predicterror_buffer[chan]);
+ av_freep(&alac->outputsamples_buffer[chan]);
+ av_freep(&alac->wasted_bits_buffer[chan]);
}
return 0;
}
-AVCodec alac_decoder = {
- "alac",
- CODEC_TYPE_AUDIO,
- CODEC_ID_ALAC,
- sizeof(ALACContext),
- alac_decode_init,
- NULL,
- alac_decode_close,
- alac_decode_frame,
+AVCodec ff_alac_decoder = {
+ .name = "alac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_ALAC,
+ .priv_data_size = sizeof(ALACContext),
+ .init = alac_decode_init,
+ .close = alac_decode_close,
+ .decode = alac_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};