* ALAC (Apple Lossless Audio Codec) decoder
* Copyright (c) 2005 David Hammerton
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
* @author 2005 David Hammerton
* @see http://crazney.net/programs/itunes/alac.html
*
- * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
+ * Note: This decoder expects a 36-byte QuickTime atom to be
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
- * bytes 0-3 atom size (0x24), big-endian
- * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
- * bytes 8-35 data bytes needed by decoder
*
- * Extradata:
- * 32bit size
- * 32bit tag (=alac)
- * 32bit zero?
- * 32bit max sample per frame
- * 8bit ?? (zero?)
+ * 32bit atom size
+ * 32bit tag ("alac")
+ * 32bit tag version (0)
+ * 32bit samples per frame (used when not set explicitly in the frames)
+ * 8bit compatible version (0)
* 8bit sample size
- * 8bit history mult
- * 8bit initial history
- * 8bit kmodifier
- * 8bit channels?
- * 16bit ??
- * 32bit max coded frame size
- * 32bit bitrate?
+ * 8bit history mult (40)
+ * 8bit initial history (14)
+ * 8bit kmodifier (10)
+ * 8bit channels
+ * 16bit maxRun (255)
+ * 32bit max coded frame size (0 means unknown)
+ * 32bit average bitrate (0 means unknown)
* 32bit samplerate
*/
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
GetBitContext gb;
int numchannels;
- int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
return x;
}
-static void bastardized_rice_decompress(ALACContext *alac,
+static int bastardized_rice_decompress(ALACContext *alac,
int32_t *output_buffer,
int output_size,
int readsamplesize, /* arg_10 */
/* standard rice encoding */
int k; /* size of extra bits */
+ if(get_bits_left(&alac->gb) <= 0)
+ return -1;
+
/* read k, that is bits as is */
k = av_log2((history >> 9) + 3);
x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
history = 0;
}
}
+ return 0;
}
static inline int sign_only(int v)
}
}
-static int alac_decode_frame(AVCodecContext *avctx,
- void *outbuffer, int *outputsize,
- AVPacket *avpkt)
+static int alac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *inbuffer = avpkt->data;
int input_buffer_size = avpkt->size;
int isnotcompressed;
uint8_t interlacing_shift;
uint8_t interlacing_leftweight;
- int i, ch;
-
- /* short-circuit null buffers */
- if (!inbuffer || !input_buffer_size)
- return -1;
+ int i, ch, ret;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
} else
outputsamples = alac->setinfo_max_samples_per_frame;
- alac->bytespersample = channels * av_get_bytes_per_sample(avctx->sample_fmt);
-
- if(outputsamples > *outputsize / alac->bytespersample){
- av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
- return -1;
+ /* get output buffer */
+ if (outputsamples > INT32_MAX) {
+ av_log(avctx, AV_LOG_ERROR, "unsupported block size: %u\n", outputsamples);
+ return AVERROR_INVALIDDATA;
+ }
+ alac->frame.nb_samples = outputsamples;
+ if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
- *outputsize = outputsamples * alac->bytespersample;
readsamplesize = alac->setinfo_sample_size - alac->extra_bits + channels - 1;
if (readsamplesize > MIN_CACHE_BITS) {
av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
if (alac->extra_bits) {
for (i = 0; i < outputsamples; i++) {
+ if(get_bits_left(&alac->gb) <= 0)
+ return -1;
for (ch = 0; ch < channels; ch++)
alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
}
}
for (ch = 0; ch < channels; ch++) {
- bastardized_rice_decompress(alac,
+ int ret = bastardized_rice_decompress(alac,
alac->predicterror_buffer[ch],
outputsamples,
readsamplesize,
alac->setinfo_rice_kmodifier,
ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
-
- if (prediction_type[ch] == 0) {
- /* adaptive fir */
- predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
- alac->outputsamples_buffer[ch],
- outputsamples,
- readsamplesize,
- predictor_coef_table[ch],
- predictor_coef_num[ch],
- prediction_quantitization[ch]);
- } else {
- av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[ch]);
- /* I think the only other prediction type (or perhaps this is
- * just a boolean?) runs adaptive fir twice.. like:
- * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
- * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
- * little strange..
+ if(ret<0)
+ return ret;
+
+ /* adaptive FIR filter */
+ if (prediction_type[ch] == 15) {
+ /* Prediction type 15 runs the adaptive FIR twice.
+ * The first pass uses the special-case coef_num = 31, while
+ * the second pass uses the coefs from the bitstream.
+ *
+ * However, this prediction type is not currently used by the
+ * reference encoder.
*/
+ predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+ alac->predicterror_buffer[ch],
+ outputsamples, readsamplesize,
+ NULL, 31, 0);
+ } else if (prediction_type[ch] > 0) {
+ av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
+ prediction_type[ch]);
}
+ predictor_decompress_fir_adapt(alac->predicterror_buffer[ch],
+ alac->outputsamples_buffer[ch],
+ outputsamples, readsamplesize,
+ predictor_coef_table[ch],
+ predictor_coef_num[ch],
+ prediction_quantitization[ch]);
}
} else {
/* not compressed, easy case */
for (i = 0; i < outputsamples; i++) {
+ if(get_bits_left(&alac->gb) <= 0)
+ return -1;
for (ch = 0; ch < channels; ch++) {
alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
alac->setinfo_sample_size);
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
- interleave_stereo_16(alac->outputsamples_buffer, outbuffer,
- outputsamples);
+ interleave_stereo_16(alac->outputsamples_buffer,
+ (int16_t *)alac->frame.data[0], outputsamples);
} else {
+ int16_t *outbuffer = (int16_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++) {
- ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i];
+ outbuffer[i] = alac->outputsamples_buffer[0][i];
}
}
break;
case 24:
if (channels == 2) {
- interleave_stereo_24(alac->outputsamples_buffer, outbuffer,
- outputsamples);
+ interleave_stereo_24(alac->outputsamples_buffer,
+ (int32_t *)alac->frame.data[0], outputsamples);
} else {
+ int32_t *outbuffer = (int32_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++)
- ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8;
+ outbuffer[i] = alac->outputsamples_buffer[0][i] << 8;
}
break;
}
if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = alac->frame;
+
return input_buffer_size;
}
ptr += 4; /* size */
ptr += 4; /* alac */
- ptr += 4; /* 0 ? */
+ ptr += 4; /* version */
if(AV_RB32(ptr) >= UINT_MAX/4){
av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
/* buffer size / 2 ? */
alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
- ptr++; /* ??? */
+ ptr++; /* compatible version */
alac->setinfo_sample_size = *ptr++;
alac->setinfo_rice_historymult = *ptr++;
alac->setinfo_rice_initialhistory = *ptr++;
alac->setinfo_rice_kmodifier = *ptr++;
alac->numchannels = *ptr++;
- bytestream_get_be16(&ptr); /* ??? */
+ bytestream_get_be16(&ptr); /* maxRun */
bytestream_get_be32(&ptr); /* max coded frame size */
- bytestream_get_be32(&ptr); /* bitrate ? */
+ bytestream_get_be32(&ptr); /* average bitrate */
bytestream_get_be32(&ptr); /* samplerate */
return 0;
return ret;
}
+ avcodec_get_frame_defaults(&alac->frame);
+ avctx->coded_frame = &alac->frame;
+
return 0;
}
.init = alac_decode_init,
.close = alac_decode_close,
.decode = alac_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff