* 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 alac.c
+ * @file
* ALAC (Apple Lossless Audio Codec) decoder
* @author 2005 David Hammerton
+ * @see http://crazney.net/programs/itunes/alac.html
*
- * For more information on the ALAC format, visit:
- * 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
*/
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "bytestream.h"
+#include "unary.h"
+#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 2
typedef struct {
AVCodecContext *avctx;
+ AVFrame frame;
GetBitContext gb;
- /* init to 0; first frame decode should initialize from extradata and
- * set this to 1 */
- int context_initialized;
- int samplesize;
int numchannels;
- int bytespersample;
/* buffers */
int32_t *predicterror_buffer[MAX_CHANNELS];
int32_t *outputsamples_buffer[MAX_CHANNELS];
+ int32_t *extra_bits_buffer[MAX_CHANNELS];
+
/* stuff from setinfo */
uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
- uint8_t setinfo_7a; /* 0x00 */
uint8_t setinfo_sample_size; /* 0x10 */
uint8_t setinfo_rice_historymult; /* 0x28 */
uint8_t setinfo_rice_initialhistory; /* 0x0a */
uint8_t setinfo_rice_kmodifier; /* 0x0e */
- uint8_t setinfo_7f; /* 0x02 */
- uint16_t setinfo_80; /* 0x00ff */
- uint32_t setinfo_82; /* 0x000020e7 */ /* max sample size?? */
- uint32_t setinfo_86; /* 0x00069fe4 */ /* bit rate (average)?? */
- uint32_t setinfo_8a_rate; /* 0x0000ac44 */
/* end setinfo stuff */
+ int extra_bits; /**< number of extra bits beyond 16-bit */
} ALACContext;
-static void allocate_buffers(ALACContext *alac)
-{
- int chan;
- for (chan = 0; chan < MAX_CHANNELS; chan++) {
- alac->predicterror_buffer[chan] =
- av_malloc(alac->setinfo_max_samples_per_frame * 4);
+static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
+ /* read x - number of 1s before 0 represent the rice */
+ int x = get_unary_0_9(gb);
- alac->outputsamples_buffer[chan] =
- av_malloc(alac->setinfo_max_samples_per_frame * 4);
- }
-}
+ if (x > 8) { /* RICE THRESHOLD */
+ /* use alternative encoding */
+ x = get_bits(gb, readsamplesize);
+ } else {
+ if (k >= limit)
+ k = limit;
-static int alac_set_info(ALACContext *alac)
-{
- unsigned char *ptr = alac->avctx->extradata;
+ if (k != 1) {
+ int extrabits = show_bits(gb, k);
- ptr += 4; /* size */
- ptr += 4; /* alac */
- ptr += 4; /* 0 ? */
+ /* multiply x by 2^k - 1, as part of their strange algorithm */
+ x = (x << k) - x;
- if(AV_RB32(ptr) >= UINT_MAX/4){
- av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
- return -1;
+ if (extrabits > 1) {
+ x += extrabits - 1;
+ skip_bits(gb, k);
+ } else
+ skip_bits(gb, k - 1);
+ }
}
-
- /* buffer size / 2 ? */
- alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
- alac->setinfo_7a = *ptr++;
- alac->setinfo_sample_size = *ptr++;
- alac->setinfo_rice_historymult = *ptr++;
- alac->setinfo_rice_initialhistory = *ptr++;
- alac->setinfo_rice_kmodifier = *ptr++;
- /* channels? */
- alac->setinfo_7f = *ptr++;
- alac->setinfo_80 = bytestream_get_be16(&ptr);
- /* max coded frame size */
- alac->setinfo_82 = bytestream_get_be32(&ptr);
- /* bitrate ? */
- alac->setinfo_86 = bytestream_get_be32(&ptr);
- /* samplerate */
- alac->setinfo_8a_rate = bytestream_get_be32(&ptr);
-
- allocate_buffers(alac);
-
- return 0;
-}
-
-static inline int count_leading_zeros(int32_t input)
-{
- return 31-av_log2(input);
+ return x;
}
static void bastardized_rice_decompress(ALACContext *alac,
int sign_modifier = 0;
for (output_count = 0; output_count < output_size; output_count++) {
- int32_t x = 0;
+ int32_t x;
int32_t x_modified;
int32_t final_val;
- /* read x - number of 1s before 0 represent the rice */
- while (x <= 8 && get_bits1(&alac->gb)) {
- x++;
- }
-
- if (x > 8) { /* RICE THRESHOLD */
- /* use alternative encoding */
- int32_t value;
-
- value = get_bits(&alac->gb, readsamplesize);
-
- /* mask value to readsamplesize size */
- if (readsamplesize != 32)
- value &= (0xffffffff >> (32 - readsamplesize));
-
- x = value;
- } else {
- /* standard rice encoding */
- int extrabits;
- int k; /* size of extra bits */
-
- /* read k, that is bits as is */
- k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3);
-
- if (k < 0)
- k += rice_kmodifier;
- else
- k = rice_kmodifier;
+ /* standard rice encoding */
+ int k; /* size of extra bits */
- if (k != 1) {
- extrabits = show_bits(&alac->gb, k);
-
- /* multiply x by 2^k - 1, as part of their strange algorithm */
- x = (x << k) - x;
-
- if (extrabits > 1) {
- x += extrabits - 1;
- skip_bits(&alac->gb, k);
- } else
- skip_bits(&alac->gb, k - 1);
- }
- }
+ /* read k, that is bits as is */
+ k = av_log2((history >> 9) + 3);
+ x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
x_modified = sign_modifier + x;
final_val = (x_modified + 1) / 2;
/* special case: there may be compressed blocks of 0 */
if ((history < 128) && (output_count+1 < output_size)) {
- int block_size;
+ int k;
+ unsigned int block_size;
sign_modifier = 1;
- x = 0;
- while (x <= 8 && get_bits1(&alac->gb)) {
- x++;
- }
-
- if (x > 8) {
- block_size = get_bits(&alac->gb, 16);
- block_size &= 0xffff;
- } else {
- int k;
- int extrabits;
-
- k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24;
-
- extrabits = show_bits(&alac->gb, k);
+ k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
- block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x
- + extrabits - 1;
-
- if (extrabits < 2) {
- x = 1 - extrabits;
- block_size += x;
- skip_bits(&alac->gb, k - 1);
- } else {
- skip_bits(&alac->gb, k);
- }
- }
+ block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
if (block_size > 0) {
+ if(block_size >= output_size - output_count){
+ av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
+ block_size= output_size - output_count - 1;
+ }
memset(&output_buffer[output_count+1], 0, block_size * 4);
output_count += block_size;
}
}
}
-static inline int32_t extend_sign32(int32_t val, int bits)
-{
- return (val << (32 - bits)) >> (32 - bits);
-}
-
static inline int sign_only(int v)
{
return v ? FFSIGN(v) : 0;
prev_value = buffer_out[i];
error_value = error_buffer[i+1];
buffer_out[i+1] =
- extend_sign32((prev_value + error_value), readsamplesize);
+ sign_extend((prev_value + error_value), readsamplesize);
}
return;
}
int32_t val;
val = buffer_out[i] + error_buffer[i+1];
- val = extend_sign32(val, readsamplesize);
+ val = sign_extend(val, readsamplesize);
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 optimised
+ * should be unrolled and optimized
*/
- if (predictor_coef_num == 4) {
- /* FIXME: optimised general case */
- return;
- }
-
- if (predictor_coef_table == 8) {
- /* FIXME: optimised general case */
- return;
- }
-#endif
/* general case */
if (predictor_coef_num > 0) {
outval = (1 << (predictor_quantitization-1)) + sum;
outval = outval >> predictor_quantitization;
outval = outval + buffer_out[0] + error_val;
- outval = extend_sign32(outval, readsamplesize);
+ outval = sign_extend(outval, readsamplesize);
buffer_out[predictor_coef_num+1] = outval;
}
}
-static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
- int16_t *buffer_out,
- int numchannels, int numsamples,
- uint8_t interlacing_shift,
- uint8_t interlacing_leftweight)
+static void decorrelate_stereo(int32_t *buffer[MAX_CHANNELS],
+ 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];
+ for (i = 0; i < numsamples; i++) {
+ int32_t a, b;
- a -= (b * interlacing_leftweight) >> interlacing_shift;
- b += a;
+ a = buffer[0][i];
+ b = buffer[1][i];
- buffer_out[i*numchannels] = b;
- buffer_out[i*numchannels + 1] = a;
- }
+ a -= (b * interlacing_leftweight) >> interlacing_shift;
+ b += a;
- return;
+ buffer[0][i] = b;
+ buffer[1][i] = a;
}
+}
+
+static void append_extra_bits(int32_t *buffer[MAX_CHANNELS],
+ int32_t *extra_bits_buffer[MAX_CHANNELS],
+ int extra_bits, int numchannels, int numsamples)
+{
+ int i, ch;
+
+ for (ch = 0; ch < numchannels; ch++)
+ for (i = 0; i < numsamples; i++)
+ buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
+}
+
+static void interleave_stereo_16(int32_t *buffer[MAX_CHANNELS],
+ int16_t *buffer_out, int numsamples)
+{
+ int i;
- /* otherwise basic interlacing took place */
for (i = 0; i < numsamples; i++) {
- int16_t left, right;
+ *buffer_out++ = buffer[0][i];
+ *buffer_out++ = buffer[1][i];
+ }
+}
- left = buffer[0][i];
- right = buffer[1][i];
+static void interleave_stereo_24(int32_t *buffer[MAX_CHANNELS],
+ int32_t *buffer_out, int numsamples)
+{
+ int i;
- buffer_out[i*numchannels] = left;
- buffer_out[i*numchannels + 1] = right;
+ for (i = 0; i < numsamples; i++) {
+ *buffer_out++ = buffer[0][i] << 8;
+ *buffer_out++ = buffer[1][i] << 8;
}
}
-static int alac_decode_frame(AVCodecContext *avctx,
- void *outbuffer, int *outputsize,
- uint8_t *inbuffer, int input_buffer_size)
+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;
ALACContext *alac = avctx->priv_data;
int channels;
- int32_t outputsamples;
+ unsigned int outputsamples;
int hassize;
- int readsamplesize;
- int wasted_bytes;
+ unsigned int readsamplesize;
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;
- }
+ int i, ch, ret;
init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
channels = get_bits(&alac->gb, 3) + 1;
- if (channels > MAX_CHANNELS) {
- av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
- MAX_CHANNELS);
- return input_buffer_size;
+ if (channels != avctx->channels) {
+ av_log(avctx, AV_LOG_ERROR, "frame header channel count mismatch\n");
+ return AVERROR_INVALIDDATA;
}
/* 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->extra_bits = get_bits(&alac->gb, 2) << 3;
/* whether the frame is compressed */
isnotcompressed = get_bits1(&alac->gb);
if (hassize) {
/* now read the number of samples as a 32bit integer */
- outputsamples = get_bits(&alac->gb, 32);
+ outputsamples = get_bits_long(&alac->gb, 32);
+ if(outputsamples > alac->setinfo_max_samples_per_frame){
+ av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
+ return -1;
+ }
} else
outputsamples = alac->setinfo_max_samples_per_frame;
- *outputsize = outputsamples * alac->bytespersample;
- readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 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;
+ }
+
+ 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);
+ 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];
- int i, chan;
+ 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];
interlacing_shift = get_bits(&alac->gb, 8);
interlacing_leftweight = get_bits(&alac->gb, 8);
- for (chan = 0; chan < channels; chan++) {
- prediction_type[chan] = get_bits(&alac->gb, 4);
- prediction_quantitization[chan] = get_bits(&alac->gb, 4);
+ for (ch = 0; ch < channels; ch++) {
+ prediction_type[ch] = get_bits(&alac->gb, 4);
+ prediction_quantitization[ch] = get_bits(&alac->gb, 4);
- ricemodifier[chan] = get_bits(&alac->gb, 3);
- predictor_coef_num[chan] = get_bits(&alac->gb, 5);
+ ricemodifier[ch] = get_bits(&alac->gb, 3);
+ predictor_coef_num[ch] = get_bits(&alac->gb, 5);
/* read the predictor table */
- for (i = 0; i < predictor_coef_num[chan]; i++)
- predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
+ for (i = 0; i < predictor_coef_num[ch]; i++)
+ predictor_coef_table[ch][i] = (int16_t)get_bits(&alac->gb, 16);
}
- if (wasted_bytes)
- av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
-
- for (chan = 0; chan < channels; chan++) {
+ if (alac->extra_bits) {
+ for (i = 0; i < outputsamples; i++) {
+ 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,
- alac->predicterror_buffer[chan],
+ alac->predicterror_buffer[ch],
outputsamples,
readsamplesize,
alac->setinfo_rice_initialhistory,
alac->setinfo_rice_kmodifier,
- ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
+ ricemodifier[ch] * alac->setinfo_rice_historymult / 4,
(1 << alac->setinfo_rice_kmodifier) - 1);
- if (prediction_type[chan] == 0) {
- /* adaptive fir */
- predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
- alac->outputsamples_buffer[chan],
- outputsamples,
- readsamplesize,
- predictor_coef_table[chan],
- predictor_coef_num[chan],
- prediction_quantitization[chan]);
- } else {
- av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
- /* 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..
+ /* 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 */
- if (alac->setinfo_sample_size <= 16) {
- int i, chan;
- for (chan = 0; chan < channels; chan++)
- for (i = 0; i < outputsamples; i++) {
- int32_t audiobits;
-
- audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
- audiobits = extend_sign32(audiobits, readsamplesize);
-
- alac->outputsamples_buffer[chan][i] = audiobits;
- }
- } else {
- int i, chan;
- for (chan = 0; chan < channels; chan++)
- for (i = 0; i < outputsamples; i++) {
- int32_t audiobits;
-
- audiobits = get_bits(&alac->gb, 16);
- /* special case of sign extension..
- * as we'll be ORing the low 16bits into this */
- audiobits = audiobits << 16;
- audiobits = audiobits >> (32 - alac->setinfo_sample_size);
- audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16);
-
- alac->outputsamples_buffer[chan][i] = audiobits;
- }
+ for (i = 0; i < outputsamples; i++) {
+ for (ch = 0; ch < channels; ch++) {
+ alac->outputsamples_buffer[ch][i] = get_sbits_long(&alac->gb,
+ alac->setinfo_sample_size);
+ }
}
- /* wasted_bytes = 0; */
+ alac->extra_bits = 0;
interlacing_shift = 0;
interlacing_leftweight = 0;
}
+ if (get_bits(&alac->gb, 3) != 7)
+ av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
+
+ if (channels == 2 && interlacing_leftweight) {
+ decorrelate_stereo(alac->outputsamples_buffer, outputsamples,
+ interlacing_shift, interlacing_leftweight);
+ }
+
+ if (alac->extra_bits) {
+ append_extra_bits(alac->outputsamples_buffer, alac->extra_bits_buffer,
+ alac->extra_bits, alac->numchannels, outputsamples);
+ }
switch(alac->setinfo_sample_size) {
case 16:
if (channels == 2) {
- reconstruct_stereo_16(alac->outputsamples_buffer,
- (int16_t*)outbuffer,
- alac->numchannels,
- outputsamples,
- interlacing_shift,
- interlacing_leftweight);
+ interleave_stereo_16(alac->outputsamples_buffer,
+ (int16_t *)alac->frame.data[0], outputsamples);
} else {
- int i;
+ int16_t *outbuffer = (int16_t *)alac->frame.data[0];
for (i = 0; i < outputsamples; i++) {
- int16_t sample = alac->outputsamples_buffer[0][i];
- ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
+ 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) {
+ 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++)
+ 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;
}
-static int alac_decode_init(AVCodecContext * avctx)
+static av_cold int alac_decode_close(AVCodecContext *avctx)
{
ALACContext *alac = avctx->priv_data;
- alac->avctx = avctx;
- alac->context_initialized = 0;
- alac->samplesize = alac->avctx->bits_per_sample;
- alac->numchannels = alac->avctx->channels;
- alac->bytespersample = (alac->samplesize / 8) * alac->numchannels;
+ int ch;
+ for (ch = 0; ch < alac->numchannels; ch++) {
+ av_freep(&alac->predicterror_buffer[ch]);
+ av_freep(&alac->outputsamples_buffer[ch]);
+ av_freep(&alac->extra_bits_buffer[ch]);
+ }
return 0;
}
-static int alac_decode_close(AVCodecContext *avctx)
+static int allocate_buffers(ALACContext *alac)
{
+ int ch;
+ for (ch = 0; ch < alac->numchannels; ch++) {
+ int buf_size = alac->setinfo_max_samples_per_frame * sizeof(int32_t);
+
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->predicterror_buffer[ch],
+ buf_size, buf_alloc_fail);
+
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->outputsamples_buffer[ch],
+ buf_size, buf_alloc_fail);
+
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->extra_bits_buffer[ch],
+ buf_size, buf_alloc_fail);
+ }
+ return 0;
+buf_alloc_fail:
+ alac_decode_close(alac->avctx);
+ return AVERROR(ENOMEM);
+}
+
+static int alac_set_info(ALACContext *alac)
+{
+ GetByteContext gb;
+
+ bytestream2_init(&gb, alac->avctx->extradata,
+ alac->avctx->extradata_size);
+
+ bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
+
+ /* buffer size / 2 ? */
+ alac->setinfo_max_samples_per_frame = bytestream2_get_be32u(&gb);
+ if (alac->setinfo_max_samples_per_frame >= UINT_MAX/4){
+ av_log(alac->avctx, AV_LOG_ERROR,
+ "setinfo_max_samples_per_frame too large\n");
+ return AVERROR_INVALIDDATA;
+ }
+ bytestream2_skipu(&gb, 1); // compatible version
+ alac->setinfo_sample_size = bytestream2_get_byteu(&gb);
+ alac->setinfo_rice_historymult = bytestream2_get_byteu(&gb);
+ alac->setinfo_rice_initialhistory = bytestream2_get_byteu(&gb);
+ alac->setinfo_rice_kmodifier = bytestream2_get_byteu(&gb);
+ alac->numchannels = bytestream2_get_byteu(&gb);
+ bytestream2_get_be16u(&gb); // maxRun
+ bytestream2_get_be32u(&gb); // max coded frame size
+ bytestream2_get_be32u(&gb); // average bitrate
+ bytestream2_get_be32u(&gb); // samplerate
+
+ return 0;
+}
+
+static av_cold int alac_decode_init(AVCodecContext * avctx)
+{
+ int ret;
ALACContext *alac = avctx->priv_data;
+ alac->avctx = avctx;
+
+ /* 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;
+ }
+
+ switch (alac->setinfo_sample_size) {
+ case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ break;
+ case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ break;
+ default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
+ alac->setinfo_sample_size);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ if (alac->numchannels < 1) {
+ av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
+ alac->numchannels = avctx->channels;
+ } else {
+ if (alac->numchannels > MAX_CHANNELS)
+ alac->numchannels = avctx->channels;
+ else
+ avctx->channels = alac->numchannels;
+ }
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
+ avctx->channels);
+ return AVERROR_PATCHWELCOME;
+ }
- int chan;
- for (chan = 0; chan < MAX_CHANNELS; chan++) {
- av_free(alac->predicterror_buffer[chan]);
- av_free(alac->outputsamples_buffer[chan]);
+ if ((ret = allocate_buffers(alac)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error allocating buffers\n");
+ return ret;
}
+ avcodec_get_frame_defaults(&alac->frame);
+ avctx->coded_frame = &alac->frame;
+
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,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff