* 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
+ * @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 rice param limit (10)
+ * 8bit channels
+ * 16bit maxRun (255)
+ * 32bit max coded frame size (0 means unknown)
+ * 32bit average bitrate (0 means unknown)
* 32bit samplerate
*/
-
+#include "libavutil/audioconvert.h"
#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "mathops.h"
#define ALAC_EXTRADATA_SIZE 36
-#define MAX_CHANNELS 2
+#define MAX_CHANNELS 8
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 numchannels;
- int bytespersample;
-
- /* buffers */
- int32_t *predicterror_buffer[MAX_CHANNELS];
-
- int32_t *outputsamples_buffer[MAX_CHANNELS];
+ int channels;
- int32_t *wasted_bits_buffer[MAX_CHANNELS];
+ int32_t *predict_error_buffer[2];
+ int32_t *output_samples_buffer[2];
+ int32_t *extra_bits_buffer[2];
- /* 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_historymult; /* 0x28 */
- uint8_t setinfo_rice_initialhistory; /* 0x0a */
- uint8_t setinfo_rice_kmodifier; /* 0x0e */
- /* end setinfo stuff */
+ uint32_t max_samples_per_frame;
+ uint8_t sample_size;
+ uint8_t rice_history_mult;
+ uint8_t rice_initial_history;
+ uint8_t rice_limit;
- int wasted_bits;
+ int extra_bits; /**< number of extra bits beyond 16-bit */
+ int nb_samples; /**< number of samples in the current frame */
} 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);
+enum RawDataBlockType {
+ /* At the moment, only SCE, CPE, LFE, and END are recognized. */
+ TYPE_SCE,
+ TYPE_CPE,
+ TYPE_CCE,
+ TYPE_LFE,
+ TYPE_DSE,
+ TYPE_PCE,
+ TYPE_FIL,
+ TYPE_END
+};
- alac->outputsamples_buffer[chan] =
- av_malloc(alac->setinfo_max_samples_per_frame * 4);
+static const uint8_t alac_channel_layout_offsets[8][8] = {
+ { 0 },
+ { 0, 1 },
+ { 2, 0, 1 },
+ { 2, 0, 1, 3 },
+ { 2, 0, 1, 3, 4 },
+ { 2, 0, 1, 4, 5, 3 },
+ { 2, 0, 1, 4, 5, 6, 3 },
+ { 2, 6, 7, 0, 1, 4, 5, 3 }
+};
- alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4);
- }
-}
+static const uint16_t alac_channel_layouts[8] = {
+ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_4POINT0,
+ AV_CH_LAYOUT_5POINT0_BACK,
+ AV_CH_LAYOUT_5POINT1_BACK,
+ AV_CH_LAYOUT_6POINT1_BACK,
+ AV_CH_LAYOUT_7POINT1_WIDE_BACK
+};
-static int alac_set_info(ALACContext *alac)
+static inline unsigned int decode_scalar(GetBitContext *gb, int k, int bps)
{
- const unsigned char *ptr = alac->avctx->extradata;
-
- ptr += 4; /* size */
- ptr += 4; /* alac */
- ptr += 4; /* 0 ? */
-
- if(AV_RB32(ptr) >= UINT_MAX/4){
- av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
- return -1;
- }
-
- /* buffer size / 2 ? */
- alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
- ptr++; /* ??? */
- alac->setinfo_sample_size = *ptr++;
- if (alac->setinfo_sample_size > 32) {
- av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
- return -1;
- }
- alac->setinfo_rice_historymult = *ptr++;
- alac->setinfo_rice_initialhistory = *ptr++;
- alac->setinfo_rice_kmodifier = *ptr++;
- ptr++; /* channels? */
- bytestream_get_be16(&ptr); /* ??? */
- bytestream_get_be32(&ptr); /* max coded frame size */
- bytestream_get_be32(&ptr); /* bitrate ? */
- bytestream_get_be32(&ptr); /* samplerate */
-
- allocate_buffers(alac);
-
- return 0;
-}
-
-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);
+ unsigned int x = get_unary_0_9(gb);
if (x > 8) { /* RICE THRESHOLD */
/* use alternative encoding */
- x = get_bits(gb, readsamplesize);
- } else {
- if (k >= limit)
- k = limit;
-
- if (k != 1) {
- int extrabits = show_bits(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(gb, k);
- } else
- skip_bits(gb, k - 1);
- }
+ x = get_bits_long(gb, bps);
+ } else if (k != 1) {
+ int extrabits = show_bits(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(gb, k);
+ } else
+ skip_bits(gb, k - 1);
}
return x;
}
-static void bastardized_rice_decompress(ALACContext *alac,
- int32_t *output_buffer,
- int output_size,
- int readsamplesize, /* arg_10 */
- int rice_initialhistory, /* arg424->b */
- int rice_kmodifier, /* arg424->d */
- int rice_historymult, /* arg424->c */
- int rice_kmodifier_mask /* arg424->e */
- )
+static void rice_decompress(ALACContext *alac, int32_t *output_buffer,
+ int nb_samples, int bps, int rice_history_mult)
{
- int output_count;
- unsigned int history = rice_initialhistory;
+ int i;
+ unsigned int history = alac->rice_initial_history;
int sign_modifier = 0;
- for (output_count = 0; output_count < output_size; output_count++) {
- int32_t x;
- int32_t x_modified;
- int32_t final_val;
-
- /* standard rice encoding */
- int k; /* size of extra bits */
+ for (i = 0; i < nb_samples; i++) {
+ int k;
+ unsigned int x;
- /* read k, that is bits as is */
+ /* calculate rice param and decode next value */
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;
- if (x_modified & 1) final_val *= -1;
-
- output_buffer[output_count] = final_val;
-
+ k = FFMIN(k, alac->rice_limit);
+ x = decode_scalar(&alac->gb, k, bps);
+ x += sign_modifier;
sign_modifier = 0;
+ output_buffer[i] = (x >> 1) ^ -(x & 1);
- /* now update the history */
- history += x_modified * rice_historymult
- - ((history * rice_historymult) >> 9);
-
- if (x_modified > 0xffff)
+ /* update the history */
+ if (x > 0xffff)
history = 0xffff;
+ else
+ history += x * rice_history_mult -
+ ((history * rice_history_mult) >> 9);
/* special case: there may be compressed blocks of 0 */
- if ((history < 128) && (output_count+1 < output_size)) {
- int k;
- unsigned int block_size;
-
- sign_modifier = 1;
-
- k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
+ if ((history < 128) && (i + 1 < nb_samples)) {
+ int block_size;
- block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
+ /* calculate rice param and decode block size */
+ k = 7 - av_log2(history) + ((history + 16) >> 6);
+ k = FFMIN(k, alac->rice_limit);
+ block_size = decode_scalar(&alac->gb, k, 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;
+ if (block_size >= nb_samples - i) {
+ av_log(alac->avctx, AV_LOG_ERROR,
+ "invalid zero block size of %d %d %d\n", block_size,
+ nb_samples, i);
+ block_size = nb_samples - i - 1;
}
- memset(&output_buffer[output_count+1], 0, block_size * 4);
- output_count += block_size;
+ memset(&output_buffer[i + 1], 0,
+ block_size * sizeof(*output_buffer));
+ i += block_size;
}
-
- if (block_size > 0xffff)
- sign_modifier = 0;
-
+ if (block_size <= 0xffff)
+ sign_modifier = 1;
history = 0;
}
}
return v ? FFSIGN(v) : 0;
}
-static void predictor_decompress_fir_adapt(int32_t *error_buffer,
- int32_t *buffer_out,
- int output_size,
- int readsamplesize,
- int16_t *predictor_coef_table,
- int predictor_coef_num,
- int predictor_quantitization)
+static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
+ int nb_samples, int bps, int16_t *lpc_coefs,
+ int lpc_order, int lpc_quant)
{
int i;
+ int32_t *pred = buffer_out;
/* first sample always copies */
*buffer_out = *error_buffer;
- if (!predictor_coef_num) {
- if (output_size <= 1)
- return;
+ if (nb_samples <= 1)
+ return;
- memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
+ if (!lpc_order) {
+ memcpy(&buffer_out[1], &error_buffer[1],
+ (nb_samples - 1) * sizeof(*buffer_out));
return;
}
- if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
- /* second-best case scenario for fir decompression,
- * error describes a small difference from the previous sample only
- */
- if (output_size <= 1)
- return;
- for (i = 0; i < output_size - 1; i++) {
- int32_t prev_value;
- int32_t error_value;
-
- prev_value = buffer_out[i];
- error_value = error_buffer[i+1];
- buffer_out[i+1] =
- sign_extend((prev_value + error_value), readsamplesize);
+ if (lpc_order == 31) {
+ /* simple 1st-order prediction */
+ for (i = 1; i < nb_samples; i++) {
+ buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i],
+ bps);
}
return;
}
/* read warm-up samples */
- if (predictor_coef_num > 0)
- for (i = 0; i < predictor_coef_num; i++) {
- int32_t val;
-
- val = buffer_out[i] + error_buffer[i+1];
- val = sign_extend(val, readsamplesize);
- buffer_out[i+1] = val;
+ for (i = 1; i <= lpc_order; i++)
+ buffer_out[i] = sign_extend(buffer_out[i - 1] + error_buffer[i], bps);
+
+ /* NOTE: 4 and 8 are very common cases that could be optimized. */
+
+ for (; i < nb_samples; i++) {
+ int j;
+ int val = 0;
+ int error_val = error_buffer[i];
+ int error_sign;
+ int d = *pred++;
+
+ /* LPC prediction */
+ for (j = 0; j < lpc_order; j++)
+ val += (pred[j] - d) * lpc_coefs[j];
+ val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
+ val += d + error_val;
+ buffer_out[i] = sign_extend(val, bps);
+
+ /* adapt LPC coefficients */
+ error_sign = sign_only(error_val);
+ if (error_sign) {
+ for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
+ int sign;
+ val = d - pred[j];
+ sign = sign_only(val) * error_sign;
+ lpc_coefs[j] -= sign;
+ val *= sign;
+ error_val -= (val >> lpc_quant) * (j + 1);
+ }
}
-
-#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) {
- for (i = predictor_coef_num + 1; i < output_size; i++) {
- int j;
- int sum = 0;
- int outval;
- int error_val = error_buffer[i];
-
- for (j = 0; j < predictor_coef_num; j++) {
- sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
- predictor_coef_table[j];
- }
-
- outval = (1 << (predictor_quantitization-1)) + sum;
- outval = outval >> predictor_quantitization;
- outval = outval + buffer_out[0] + error_val;
- outval = sign_extend(outval, readsamplesize);
-
- buffer_out[predictor_coef_num+1] = outval;
-
- if (error_val > 0) {
- int predictor_num = predictor_coef_num - 1;
-
- while (predictor_num >= 0 && error_val > 0) {
- int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
- int sign = sign_only(val);
-
- predictor_coef_table[predictor_num] -= sign;
-
- val *= sign; /* absolute value */
-
- error_val -= ((val >> predictor_quantitization) *
- (predictor_coef_num - predictor_num));
-
- predictor_num--;
- }
- } else if (error_val < 0) {
- int predictor_num = predictor_coef_num - 1;
-
- while (predictor_num >= 0 && error_val < 0) {
- int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
- int sign = - sign_only(val);
+}
- predictor_coef_table[predictor_num] -= sign;
+static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
+ int decorr_shift, int decorr_left_weight)
+{
+ int i;
- val *= sign; /* neg value */
+ for (i = 0; i < nb_samples; i++) {
+ int32_t a, b;
- error_val -= ((val >> predictor_quantitization) *
- (predictor_coef_num - predictor_num));
+ a = buffer[0][i];
+ b = buffer[1][i];
- predictor_num--;
- }
- }
+ a -= (b * decorr_left_weight) >> decorr_shift;
+ b += a;
- buffer_out++;
- }
+ buffer[0][i] = b;
+ buffer[1][i] = a;
}
}
-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 append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
+ int extra_bits, int channels, int nb_samples)
{
- int i;
- if (numsamples <= 0)
- return;
+ int i, ch;
- /* weighted interlacing */
- if (interlacing_leftweight) {
- for (i = 0; i < numsamples; i++) {
- int32_t a, b;
+ for (ch = 0; ch < channels; ch++)
+ for (i = 0; i < nb_samples; i++)
+ buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
+}
- a = buffer[0][i];
- b = buffer[1][i];
+static int decode_element(AVCodecContext *avctx, void *data, int ch_index,
+ int channels)
+{
+ ALACContext *alac = avctx->priv_data;
+ int has_size, bps, is_compressed, decorr_shift, decorr_left_weight, ret;
+ uint32_t output_samples;
+ int i, ch;
- a -= (b * interlacing_leftweight) >> interlacing_shift;
- b += a;
+ skip_bits(&alac->gb, 4); /* element instance tag */
+ skip_bits(&alac->gb, 12); /* unused header bits */
- buffer_out[i*numchannels] = b;
- buffer_out[i*numchannels + 1] = a;
- }
+ /* the number of output samples is stored in the frame */
+ has_size = get_bits1(&alac->gb);
- return;
+ alac->extra_bits = get_bits(&alac->gb, 2) << 3;
+ bps = alac->sample_size - alac->extra_bits + channels - 1;
+ if (bps > 32) {
+ av_log(avctx, AV_LOG_ERROR, "bps is unsupported: %d\n", bps);
+ return AVERROR_PATCHWELCOME;
}
- /* otherwise basic interlacing took place */
- for (i = 0; i < numsamples; i++) {
- int16_t left, right;
-
- left = buffer[0][i];
- right = buffer[1][i];
-
- buffer_out[i*numchannels] = left;
- buffer_out[i*numchannels + 1] = right;
+ /* whether the frame is compressed */
+ is_compressed = !get_bits1(&alac->gb);
+
+ if (has_size)
+ output_samples = get_bits_long(&alac->gb, 32);
+ else
+ output_samples = alac->max_samples_per_frame;
+ if (!output_samples || output_samples > alac->max_samples_per_frame) {
+ av_log(avctx, AV_LOG_ERROR, "invalid samples per frame: %d\n",
+ output_samples);
+ return AVERROR_INVALIDDATA;
+ }
+ if (!alac->nb_samples) {
+ /* get output buffer */
+ alac->frame.nb_samples = output_samples;
+ if ((ret = avctx->get_buffer(avctx, &alac->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ } else if (output_samples != alac->nb_samples) {
+ av_log(avctx, AV_LOG_ERROR, "sample count mismatch: %u != %d\n",
+ output_samples, alac->nb_samples);
+ return AVERROR_INVALIDDATA;
+ }
+ alac->nb_samples = output_samples;
+ if (alac->sample_size > 16) {
+ for (ch = 0; ch < channels; ch++)
+ alac->output_samples_buffer[ch] = (int32_t *)alac->frame.extended_data[ch_index + ch];
}
-}
-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 (is_compressed) {
+ int16_t lpc_coefs[2][32];
+ int lpc_order[2];
+ int prediction_type[2];
+ int lpc_quant[2];
+ int rice_history_mult[2];
- if (numsamples <= 0)
- return;
+ decorr_shift = get_bits(&alac->gb, 8);
+ decorr_left_weight = get_bits(&alac->gb, 8);
- /* weighted interlacing */
- if (interlacing_leftweight) {
- for (i = 0; i < numsamples; i++) {
- int32_t a, b;
+ for (ch = 0; ch < channels; ch++) {
+ prediction_type[ch] = get_bits(&alac->gb, 4);
+ lpc_quant[ch] = get_bits(&alac->gb, 4);
+ rice_history_mult[ch] = get_bits(&alac->gb, 3);
+ lpc_order[ch] = get_bits(&alac->gb, 5);
- a = buffer[0][i];
- b = buffer[1][i];
-
- a -= (b * interlacing_leftweight) >> interlacing_shift;
- b += a;
+ /* read the predictor table */
+ for (i = lpc_order[ch] - 1; i >= 0; i--)
+ lpc_coefs[ch][i] = get_sbits(&alac->gb, 16);
+ }
- if (wasted_bits) {
- b = (b << wasted_bits) | wasted_bits_buffer[0][i];
- a = (a << wasted_bits) | wasted_bits_buffer[1][i];
+ if (alac->extra_bits) {
+ for (i = 0; i < alac->nb_samples; i++) {
+ for (ch = 0; ch < channels; ch++)
+ alac->extra_bits_buffer[ch][i] = get_bits(&alac->gb, alac->extra_bits);
}
-
- buffer_out[i * numchannels] = b << 8;
- buffer_out[i * numchannels + 1] = a << 8;
+ }
+ for (ch = 0; ch < channels; ch++) {
+ rice_decompress(alac, alac->predict_error_buffer[ch],
+ alac->nb_samples, bps,
+ rice_history_mult[ch] * alac->rice_history_mult / 4);
+
+ /* 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.
+ */
+ lpc_prediction(alac->predict_error_buffer[ch],
+ alac->predict_error_buffer[ch],
+ alac->nb_samples, bps, NULL, 31, 0);
+ } else if (prediction_type[ch] > 0) {
+ av_log(avctx, AV_LOG_WARNING, "unknown prediction type: %i\n",
+ prediction_type[ch]);
+ }
+ lpc_prediction(alac->predict_error_buffer[ch],
+ alac->output_samples_buffer[ch], alac->nb_samples,
+ bps, lpc_coefs[ch], lpc_order[ch], lpc_quant[ch]);
}
} else {
- for (i = 0; i < numsamples; i++) {
- int32_t left, right;
+ /* not compressed, easy case */
+ for (i = 0; i < alac->nb_samples; i++) {
+ for (ch = 0; ch < channels; ch++) {
+ alac->output_samples_buffer[ch][i] =
+ get_sbits_long(&alac->gb, alac->sample_size);
+ }
+ }
+ alac->extra_bits = 0;
+ decorr_shift = 0;
+ decorr_left_weight = 0;
+ }
- left = buffer[0][i];
- right = buffer[1][i];
+ if (channels == 2 && decorr_left_weight) {
+ decorrelate_stereo(alac->output_samples_buffer, alac->nb_samples,
+ decorr_shift, decorr_left_weight);
+ }
- if (wasted_bits) {
- left = (left << wasted_bits) | wasted_bits_buffer[0][i];
- right = (right << wasted_bits) | wasted_bits_buffer[1][i];
- }
+ if (alac->extra_bits) {
+ append_extra_bits(alac->output_samples_buffer, alac->extra_bits_buffer,
+ alac->extra_bits, channels, alac->nb_samples);
+ }
- buffer_out[i * numchannels] = left << 8;
- buffer_out[i * numchannels + 1] = right << 8;
- }
+ switch(alac->sample_size) {
+ case 16: {
+ for (ch = 0; ch < channels; ch++) {
+ int16_t *outbuffer = (int16_t *)alac->frame.extended_data[ch_index + ch];
+ for (i = 0; i < alac->nb_samples; i++)
+ *outbuffer++ = alac->output_samples_buffer[ch][i];
+ }}
+ break;
+ case 24: {
+ for (ch = 0; ch < channels; ch++) {
+ for (i = 0; i < alac->nb_samples; i++)
+ alac->output_samples_buffer[ch][i] <<= 8;
+ }}
+ break;
}
+
+ return 0;
}
-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;
ALACContext *alac = avctx->priv_data;
-
+ enum RawDataBlockType element;
int channels;
- unsigned int outputsamples;
- int hassize;
- 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;
+ int ch, ret, got_end;
+
+ init_get_bits(&alac->gb, avpkt->data, avpkt->size * 8);
+
+ got_end = 0;
+ alac->nb_samples = 0;
+ ch = 0;
+ while (get_bits_left(&alac->gb) >= 3) {
+ element = get_bits(&alac->gb, 3);
+ if (element == TYPE_END) {
+ got_end = 1;
+ break;
}
- if (alac_set_info(alac)) {
- av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
- return input_buffer_size;
+ if (element > TYPE_CPE && element != TYPE_LFE) {
+ av_log(avctx, AV_LOG_ERROR, "syntax element unsupported: %d", element);
+ return AVERROR_PATCHWELCOME;
}
- alac->context_initialized = 1;
- }
- 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;
- }
+ channels = (element == TYPE_CPE) ? 2 : 1;
+ if (ch + channels > alac->channels) {
+ av_log(avctx, AV_LOG_ERROR, "invalid element channel count\n");
+ return AVERROR_INVALIDDATA;
+ }
- /* 2^result = something to do with output waiting.
- * perhaps matters if we read > 1 frame in a pass?
- */
- skip_bits(&alac->gb, 4);
+ ret = decode_element(avctx, data,
+ alac_channel_layout_offsets[alac->channels - 1][ch],
+ channels);
+ if (ret < 0 && get_bits_left(&alac->gb))
+ return ret;
- skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
+ ch += channels;
+ }
+ if (!got_end) {
+ av_log(avctx, AV_LOG_ERROR, "no end tag found. incomplete packet.\n");
+ return AVERROR_INVALIDDATA;
+ }
- /* the output sample size is stored soon */
- hassize = get_bits1(&alac->gb);
+ if (avpkt->size * 8 - get_bits_count(&alac->gb) > 8) {
+ av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n",
+ avpkt->size * 8 - get_bits_count(&alac->gb));
+ }
- alac->wasted_bits = get_bits(&alac->gb, 2) << 3;
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = alac->frame;
- /* 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_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;
+ return avpkt->size;
+}
- switch (alac->setinfo_sample_size) {
- case 16: avctx->sample_fmt = SAMPLE_FMT_S16;
- alac->bytespersample = channels << 1;
- break;
- case 24: avctx->sample_fmt = 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;
- }
+static av_cold int alac_decode_close(AVCodecContext *avctx)
+{
+ ALACContext *alac = avctx->priv_data;
- if(outputsamples > *outputsize / alac->bytespersample){
- av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
- return -1;
+ int ch;
+ for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
+ av_freep(&alac->predict_error_buffer[ch]);
+ if (alac->sample_size == 16)
+ av_freep(&alac->output_samples_buffer[ch]);
+ av_freep(&alac->extra_bits_buffer[ch]);
}
- *outputsize = outputsamples * alac->bytespersample;
- 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;
- }
+ return 0;
+}
- 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;
+static int allocate_buffers(ALACContext *alac)
+{
+ int ch;
+ int buf_size = alac->max_samples_per_frame * sizeof(int32_t);
- interlacing_shift = get_bits(&alac->gb, 8);
- interlacing_leftweight = get_bits(&alac->gb, 8);
+ for (ch = 0; ch < FFMIN(alac->channels, 2); ch++) {
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->predict_error_buffer[ch],
+ buf_size, buf_alloc_fail);
- for (chan = 0; chan < channels; chan++) {
- prediction_type[chan] = get_bits(&alac->gb, 4);
- prediction_quantitization[chan] = get_bits(&alac->gb, 4);
+ if (alac->sample_size == 16) {
+ FF_ALLOC_OR_GOTO(alac->avctx, alac->output_samples_buffer[ch],
+ buf_size, buf_alloc_fail);
+ }
- ricemodifier[chan] = get_bits(&alac->gb, 3);
- predictor_coef_num[chan] = get_bits(&alac->gb, 5);
+ 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);
+}
- /* 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);
- }
+static int alac_set_info(ALACContext *alac)
+{
+ GetByteContext gb;
- 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],
- outputsamples,
- readsamplesize,
- alac->setinfo_rice_initialhistory,
- alac->setinfo_rice_kmodifier,
- ricemodifier[chan] * 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..
- */
- }
- }
- } 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;
+ bytestream2_init(&gb, alac->avctx->extradata,
+ alac->avctx->extradata_size);
- audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);
+ bytestream2_skipu(&gb, 12); // size:4, alac:4, version:4
- alac->outputsamples_buffer[chan][i] = audiobits;
- }
- } 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;
- }
- if (get_bits(&alac->gb, 3) != 7)
- av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
-
- 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);
- } else {
- int i;
- for (i = 0; i < outputsamples; i++) {
- int16_t sample = alac->outputsamples_buffer[0][i];
- ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
- }
- }
- break;
- case 24:
- 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;
+ alac->max_samples_per_frame = bytestream2_get_be32u(&gb);
+ if (!alac->max_samples_per_frame || alac->max_samples_per_frame > INT_MAX) {
+ av_log(alac->avctx, AV_LOG_ERROR, "max samples per frame invalid: %u\n",
+ alac->max_samples_per_frame);
+ return AVERROR_INVALIDDATA;
}
+ bytestream2_skipu(&gb, 1); // compatible version
+ alac->sample_size = bytestream2_get_byteu(&gb);
+ alac->rice_history_mult = bytestream2_get_byteu(&gb);
+ alac->rice_initial_history = bytestream2_get_byteu(&gb);
+ alac->rice_limit = bytestream2_get_byteu(&gb);
+ alac->channels = 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
- 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));
-
- return input_buffer_size;
+ return 0;
}
static av_cold int alac_decode_init(AVCodecContext * avctx)
{
+ int ret;
ALACContext *alac = avctx->priv_data;
alac->avctx = avctx;
- alac->context_initialized = 0;
- alac->numchannels = alac->avctx->channels;
+ /* 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;
-}
+ switch (alac->sample_size) {
+ case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ break;
+ case 24:
+ case 32: avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ break;
+ default: av_log_ask_for_sample(avctx, "Sample depth %d is not supported.\n",
+ alac->sample_size);
+ return AVERROR_PATCHWELCOME;
+ }
+ avctx->bits_per_raw_sample = alac->sample_size;
-static av_cold int alac_decode_close(AVCodecContext *avctx)
-{
- ALACContext *alac = avctx->priv_data;
+ if (alac->channels < 1) {
+ av_log(avctx, AV_LOG_WARNING, "Invalid channel count\n");
+ alac->channels = avctx->channels;
+ } else {
+ if (alac->channels > MAX_CHANNELS)
+ alac->channels = avctx->channels;
+ else
+ avctx->channels = alac->channels;
+ }
+ if (avctx->channels > MAX_CHANNELS) {
+ av_log(avctx, AV_LOG_ERROR, "Unsupported channel count: %d\n",
+ avctx->channels);
+ return AVERROR_PATCHWELCOME;
+ }
+ avctx->channel_layout = alac_channel_layouts[alac->channels - 1];
- int chan;
- for (chan = 0; chan < MAX_CHANNELS; chan++) {
- av_free(alac->predicterror_buffer[chan]);
- av_free(alac->outputsamples_buffer[chan]);
- av_freep(&alac->wasted_bits_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,
- .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
+AVCodec ff_alac_decoder = {
+ .name = "alac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_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