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authorMike Melanson <mike@multimedia.cx>2005-03-06 00:43:55 +0000
committerMike Melanson <mike@multimedia.cx>2005-03-06 00:43:55 +0000
commit6d6d7970e7c0ae1bf3f0e015d3c22723ed5b1a28 (patch)
treed31dea6d1e1a23066c169eecaa91d3eb8bc279fd
parent2a515c08f25dc4c78c09b4600358574ef34b4ea5 (diff)
downloadffmpeg-streaming-6d6d7970e7c0ae1bf3f0e015d3c22723ed5b1a28.zip
ffmpeg-streaming-6d6d7970e7c0ae1bf3f0e015d3c22723ed5b1a28.tar.gz
first pass at ALAC decoder from David Hammerton; while David's original
decoder works great, this decoder is not completely and seamlessly integrated yet with FFmpeg Originally committed as revision 4008 to svn://svn.ffmpeg.org/ffmpeg/trunk
-rw-r--r--CREDITS1
-rw-r--r--libavcodec/Makefile5
-rw-r--r--libavcodec/alac.c970
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/avcodec.h4
-rw-r--r--libavformat/mov.c18
6 files changed, 996 insertions, 3 deletions
diff --git a/CREDITS b/CREDITS
index 41bb8b9..dfcaa55 100644
--- a/CREDITS
+++ b/CREDITS
@@ -14,6 +14,7 @@ Brian Foley
Arpad Gereoffy
Philip Gladstone
Vladimir Gneushev
+David Hammerton
Wolfgang Hesseler
Falk Hueffner
Zdenek Kabelac
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 0ee56e6..25c7adf 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -1,6 +1,6 @@
#
# libavcodec Makefile
-# (c) 2000-2003 Fabrice Bellard
+# (c) 2000-2005 Fabrice Bellard
#
include ../config.mak
@@ -22,7 +22,8 @@ OBJS= bitstream.o utils.o mem.o allcodecs.o \
smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o \
flac.o vp3dsp.o integer.o snow.o tscc.o sonic.o ulti.o h264idct.o \
qdrw.o xl.o rangecoder.o png.o pnm.o qpeg.o vc9.o h263.o h261.o \
- msmpeg4.o h263dec.o svq1.o rv10.o wmadec.o indeo3.o shorten.o loco.o
+ msmpeg4.o h263dec.o svq1.o rv10.o wmadec.o indeo3.o shorten.o loco.o \
+ alac.o
AMROBJS=
ifeq ($(AMR_NB),yes)
diff --git a/libavcodec/alac.c b/libavcodec/alac.c
new file mode 100644
index 0000000..05237073
--- /dev/null
+++ b/libavcodec/alac.c
@@ -0,0 +1,970 @@
+/*
+ * ALAC (Apple Lossless Audio Codec) decoder
+ * Copyright (c) 2005 David Hammerton
+ * All rights reserved.
+ *
+ * This library 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 of the License, or (at your option) any later version.
+ *
+ * This library 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 this library; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+/**
+ * @file alac.c
+ * 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
+ *
+ * Note: This decoder expects a 36- (0x24-)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
+ */
+
+
+#include "avcodec.h"
+
+#define ALAC_EXTRADATA_SIZE 36
+
+struct alac_file {
+ unsigned char *input_buffer;
+ int input_buffer_index;
+ int input_buffer_size;
+ int input_buffer_bitaccumulator; /* used so we can do arbitary
+ bit reads */
+
+ int samplesize;
+ int numchannels;
+ int bytespersample;
+
+
+ /* buffers */
+ int32_t *predicterror_buffer_a;
+ int32_t *predicterror_buffer_b;
+
+ int32_t *outputsamples_buffer_a;
+ int32_t *outputsamples_buffer_b;
+
+
+ /* 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 */
+ uint32_t setinfo_86; /* 0x00069fe4 */
+ uint32_t setinfo_8a_rate; /* 0x0000ac44 */
+ /* end setinfo stuff */
+};
+
+typedef struct alac_file alac_file;
+
+typedef struct {
+
+ AVCodecContext *avctx;
+ /* init to 0; first frame decode should initialize from extradata and
+ * set this to 1 */
+ int context_initialized;
+
+ alac_file *alac;
+} ALACContext;
+
+static void allocate_buffers(alac_file *alac)
+{
+ alac->predicterror_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+ alac->predicterror_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+
+ alac->outputsamples_buffer_a = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+ alac->outputsamples_buffer_b = av_malloc(alac->setinfo_max_samples_per_frame * 4);
+}
+
+void alac_set_info(alac_file *alac, char *inputbuffer)
+{
+ char *ptr = inputbuffer;
+
+ ptr += 4; /* size */
+ ptr += 4; /* alac */
+ ptr += 4; /* 0 ? */
+
+ alac->setinfo_max_samples_per_frame = BE_32(ptr); /* buffer size / 2 ? */
+ ptr += 4;
+ alac->setinfo_7a = *ptr++;
+ alac->setinfo_sample_size = *ptr++;
+ alac->setinfo_rice_historymult = *ptr++;
+ alac->setinfo_rice_initialhistory = *ptr++;
+ alac->setinfo_rice_kmodifier = *ptr++;
+ alac->setinfo_7f = *ptr++;
+ alac->setinfo_80 = BE_16(ptr);
+ ptr += 2;
+ alac->setinfo_82 = BE_32(ptr);
+ ptr += 4;
+ alac->setinfo_86 = BE_32(ptr);
+ ptr += 4;
+ alac->setinfo_8a_rate = BE_32(ptr);
+ ptr += 4;
+
+ allocate_buffers(alac);
+}
+
+/* stream reading */
+
+/* supports reading 1 to 16 bits, in big endian format */
+static uint32_t readbits_16(alac_file *alac, int bits)
+{
+ uint32_t result;
+ int new_accumulator;
+
+ if (alac->input_buffer_index + 2 >= alac->input_buffer_size) {
+ av_log(NULL, AV_LOG_INFO, "alac: input buffer went out of bounds (%d >= %d)\n",
+ alac->input_buffer_index + 2, alac->input_buffer_size);
+ exit (0);
+ }
+ result = (alac->input_buffer[alac->input_buffer_index + 0] << 16) |
+ (alac->input_buffer[alac->input_buffer_index + 1] << 8) |
+ (alac->input_buffer[alac->input_buffer_index + 2]);
+
+ /* shift left by the number of bits we've already read,
+ * so that the top 'n' bits of the 24 bits we read will
+ * be the return bits */
+ result = result << alac->input_buffer_bitaccumulator;
+
+ result = result & 0x00ffffff;
+
+ /* and then only want the top 'n' bits from that, where
+ * n is 'bits' */
+ result = result >> (24 - bits);
+
+ new_accumulator = (alac->input_buffer_bitaccumulator + bits);
+
+ /* increase the buffer pointer if we've read over n bytes. */
+ alac->input_buffer_index += (new_accumulator >> 3);
+
+ /* and the remainder goes back into the bit accumulator */
+ alac->input_buffer_bitaccumulator = (new_accumulator & 7);
+
+ return result;
+}
+
+/* supports reading 1 to 32 bits, in big endian format */
+static uint32_t readbits(alac_file *alac, int bits)
+{
+ int32_t result = 0;
+
+ if (bits > 16) {
+ bits -= 16;
+ result = readbits_16(alac, 16) << bits;
+ }
+
+ result |= readbits_16(alac, bits);
+
+ return result;
+}
+
+/* reads a single bit */
+static int readbit(alac_file *alac)
+{
+ int result;
+ int new_accumulator;
+
+ if (alac->input_buffer_index >= alac->input_buffer_size) {
+ av_log(NULL, AV_LOG_INFO, "alac: input buffer went out of bounds (%d >= %d)\n",
+ alac->input_buffer_index + 2, alac->input_buffer_size);
+ exit (0);
+ }
+
+ result = alac->input_buffer[alac->input_buffer_index];
+
+ result = result << alac->input_buffer_bitaccumulator;
+
+ result = result >> 7 & 1;
+
+ new_accumulator = (alac->input_buffer_bitaccumulator + 1);
+
+ alac->input_buffer_index += (new_accumulator / 8);
+
+ alac->input_buffer_bitaccumulator = (new_accumulator % 8);
+
+ return result;
+}
+
+static void unreadbits(alac_file *alac, int bits)
+{
+ int new_accumulator = (alac->input_buffer_bitaccumulator - bits);
+
+ alac->input_buffer_index += (new_accumulator >> 3);
+
+ alac->input_buffer_bitaccumulator = (new_accumulator & 7);
+ if (alac->input_buffer_bitaccumulator < 0)
+ alac->input_buffer_bitaccumulator *= -1;
+}
+
+/* hideously inefficient. could use a bitmask search,
+ * alternatively bsr on x86,
+ */
+static int count_leading_zeros(int32_t input)
+{
+ int i = 0;
+ while (!(0x80000000 & input) && i < 32) {
+ i++;
+ input = input << 1;
+ }
+ return i;
+}
+
+void bastardized_rice_decompress(alac_file *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 */
+ )
+{
+ int output_count;
+ unsigned int history = rice_initialhistory;
+ int sign_modifier = 0;
+
+ for (output_count = 0; output_count < output_size; output_count++) {
+ int32_t x = 0;
+ int32_t x_modified;
+ int32_t final_val;
+
+ /* read x - number of 1s before 0 represent the rice */
+ while (x <= 8 && readbit(alac)) {
+ x++;
+ }
+
+
+ if (x > 8) { /* RICE THRESHOLD */
+ /* use alternative encoding */
+ int32_t value;
+
+ value = readbits(alac, 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;
+
+ if (k != 1) {
+ extrabits = readbits(alac, k);
+
+ /* multiply x by 2^k - 1, as part of their strange algorithm */
+ x = (x << k) - x;
+
+ if (extrabits > 1) {
+ x += extrabits - 1;
+ } else
+ unreadbits(alac, 1);
+ }
+ }
+
+ x_modified = sign_modifier + x;
+ final_val = (x_modified + 1) / 2;
+ if (x_modified & 1) final_val *= -1;
+
+ output_buffer[output_count] = final_val;
+
+ sign_modifier = 0;
+
+ /* now update the history */
+ history += (x_modified * rice_historymult)
+ - ((history * rice_historymult) >> 9);
+
+ if (x_modified > 0xffff)
+ history = 0xffff;
+
+ /* special case: there may be compressed blocks of 0 */
+ if ((history < 128) && (output_count+1 < output_size)) {
+ int block_size;
+
+ sign_modifier = 1;
+
+ x = 0;
+ while (x <= 8 && readbit(alac)) {
+ x++;
+ }
+
+ if (x > 8) {
+ block_size = readbits(alac, 16);
+ block_size &= 0xffff;
+ } else {
+ int k;
+ int extrabits;
+
+ k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24;
+
+ extrabits = readbits(alac, k);
+
+ block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x
+ + extrabits - 1;
+
+ if (extrabits < 2) {
+ x = 1 - extrabits;
+ block_size += x;
+ unreadbits(alac, 1);
+ }
+ }
+
+ if (block_size > 0) {
+ memset(&output_buffer[output_count+1], 0, block_size * 4);
+ output_count += block_size;
+
+ }
+
+ if (block_size > 0xffff)
+ sign_modifier = 0;
+
+ history = 0;
+ }
+ }
+}
+
+#define SIGN_EXTENDED32(val, bits) ((val << (32 - bits)) >> (32 - bits))
+
+#define SIGN_ONLY(v) \
+ ((v < 0) ? (-1) : \
+ ((v > 0) ? (1) : \
+ (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)
+{
+ int i;
+
+ /* first sample always copies */
+ *buffer_out = *error_buffer;
+
+ if (!predictor_coef_num) {
+ if (output_size <= 1) return;
+ memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
+ 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_EXTENDED32((prev_value + error_value), readsamplesize);
+ }
+ return;
+ }
+
+ /* read warm-up samples */
+ if (predictor_coef_num > 0) {
+ int i;
+ for (i = 0; i < predictor_coef_num; i++) {
+ int32_t val;
+
+ val = buffer_out[i] + error_buffer[i+1];
+
+ val = SIGN_EXTENDED32(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
+ */
+ 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) {
+ 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_EXTENDED32(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;
+
+ val *= sign; /* neg value */
+
+ error_val -= ((val >> predictor_quantitization) *
+ (predictor_coef_num - predictor_num));
+
+ predictor_num--;
+ }
+ }
+
+ buffer_out++;
+ }
+ }
+}
+
+void deinterlace_16(int32_t *buffer_a, int32_t *buffer_b,
+ int16_t *buffer_out,
+ 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 difference, midright;
+ int16_t left;
+ int16_t right;
+
+ midright = buffer_a[i];
+ difference = buffer_b[i];
+
+
+ right = midright - ((difference * interlacing_leftweight) >> interlacing_shift);
+ left = (midright - ((difference * interlacing_leftweight) >> interlacing_shift))
+ + difference;
+
+ /* output is always little endian */
+/*
+ if (host_bigendian) {
+ be2me_16(left);
+ be2me_16(right);
+ }
+*/
+
+ buffer_out[i*numchannels] = left;
+ buffer_out[i*numchannels + 1] = right;
+ }
+
+ return;
+ }
+
+ /* otherwise basic interlacing took place */
+ for (i = 0; i < numsamples; i++) {
+ int16_t left, right;
+
+ left = buffer_a[i];
+ right = buffer_b[i];
+
+ /* output is always little endian */
+/*
+ if (host_bigendian) {
+ be2me_16(left);
+ be2me_16(right);
+ }
+*/
+
+ buffer_out[i*numchannels] = left;
+ buffer_out[i*numchannels + 1] = right;
+ }
+}
+
+int decode_frame(ALACContext *s, alac_file *alac,
+ unsigned char *inbuffer,
+ int input_buffer_size,
+ void *outbuffer, int *outputsize){
+
+ int channels;
+ int32_t outputsamples = alac->setinfo_max_samples_per_frame;
+
+ /* initialize from the extradata */
+ if (!s->context_initialized) {
+ if (s->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
+ av_log(NULL, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
+ ALAC_EXTRADATA_SIZE);
+ return input_buffer_size;
+ }
+ alac_set_info(s->alac, s->avctx->extradata);
+ s->context_initialized = 1;
+ }
+
+
+ /* setup the stream */
+ alac->input_buffer = inbuffer;
+ alac->input_buffer_index = 0;
+ alac->input_buffer_size = input_buffer_size;
+ alac->input_buffer_bitaccumulator = 0;
+
+ channels = readbits(alac, 3);
+
+ *outputsize = outputsamples * alac->bytespersample;
+
+ switch(channels) {
+ case 0: { /* 1 channel */
+ int hassize;
+ int isnotcompressed;
+ int readsamplesize;
+
+ int wasted_bytes;
+ int ricemodifier;
+
+
+ /* 2^result = something to do with output waiting.
+ * perhaps matters if we read > 1 frame in a pass?
+ */
+ readbits(alac, 4);
+
+ readbits(alac, 12); /* unknown, skip 12 bits */
+
+ hassize = readbits(alac, 1); /* the output sample size is stored soon */
+
+ wasted_bytes = readbits(alac, 2); /* unknown ? */
+
+ isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
+
+ if (hassize) {
+ /* now read the number of samples,
+ * as a 32bit integer */
+ outputsamples = readbits(alac, 32);
+ *outputsize = outputsamples * alac->bytespersample;
+ }
+
+ readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8);
+
+ if (!isnotcompressed) {
+ /* so it is compressed */
+ int16_t predictor_coef_table[32];
+ int predictor_coef_num;
+ int prediction_type;
+ int prediction_quantitization;
+ int i;
+
+ /* skip 16 bits, not sure what they are. seem to be used in
+ * two channel case */
+ readbits(alac, 8);
+ readbits(alac, 8);
+
+ prediction_type = readbits(alac, 4);
+ prediction_quantitization = readbits(alac, 4);
+
+ ricemodifier = readbits(alac, 3);
+ predictor_coef_num = readbits(alac, 5);
+
+ /* read the predictor table */
+ for (i = 0; i < predictor_coef_num; i++) {
+ predictor_coef_table[i] = (int16_t)readbits(alac, 16);
+ }
+
+ if (wasted_bytes) {
+ /* these bytes seem to have something to do with
+ * > 2 channel files.
+ */
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
+ }
+
+ bastardized_rice_decompress(alac,
+ alac->predicterror_buffer_a,
+ outputsamples,
+ readsamplesize,
+ alac->setinfo_rice_initialhistory,
+ alac->setinfo_rice_kmodifier,
+ ricemodifier * alac->setinfo_rice_historymult / 4,
+ (1 << alac->setinfo_rice_kmodifier) - 1);
+
+ if (prediction_type == 0) {
+ /* adaptive fir */
+ predictor_decompress_fir_adapt(alac->predicterror_buffer_a,
+ alac->outputsamples_buffer_a,
+ outputsamples,
+ readsamplesize,
+ predictor_coef_table,
+ predictor_coef_num,
+ prediction_quantitization);
+ } else {
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type);
+ /* 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 */
+ if (readsamplesize <= 16) {
+ int i;
+ for (i = 0; i < outputsamples; i++) {
+ int32_t audiobits = readbits(alac, readsamplesize);
+
+ audiobits = SIGN_EXTENDED32(audiobits, readsamplesize);
+
+ alac->outputsamples_buffer_a[i] = audiobits;
+ }
+ } else {
+ int i;
+ for (i = 0; i < outputsamples; i++) {
+ int32_t audiobits;
+
+ audiobits = readbits(alac, 16);
+ /* special case of sign extension..
+ * as we'll be ORing the low 16bits into this */
+ audiobits = audiobits << 16;
+ audiobits = audiobits >> (32 - readsamplesize);
+
+ audiobits |= readbits(alac, readsamplesize - 16);
+
+ alac->outputsamples_buffer_a[i] = audiobits;
+ }
+ }
+ /* wasted_bytes = 0; // unused */
+ }
+
+ switch(alac->setinfo_sample_size) {
+ case 16: {
+ int i;
+ for (i = 0; i < outputsamples; i++) {
+ int16_t sample = alac->outputsamples_buffer_a[i];
+ be2me_16(sample);
+ ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
+ }
+ break;
+ }
+ case 20:
+ case 24:
+ case 32:
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case 1: { /* 2 channels */
+ int hassize;
+ int isnotcompressed;
+ int readsamplesize;
+
+ int wasted_bytes;
+
+ uint8_t interlacing_shift;
+ uint8_t interlacing_leftweight;
+
+ /* 2^result = something to do with output waiting.
+ * perhaps matters if we read > 1 frame in a pass?
+ */
+ readbits(alac, 4);
+
+ readbits(alac, 12); /* unknown, skip 12 bits */
+
+ hassize = readbits(alac, 1); /* the output sample size is stored soon */
+
+ wasted_bytes = readbits(alac, 2); /* unknown ? */
+
+ isnotcompressed = readbits(alac, 1); /* whether the frame is compressed */
+
+ if (hassize) {
+ /* now read the number of samples,
+ * as a 32bit integer */
+ outputsamples = readbits(alac, 32);
+ *outputsize = outputsamples * alac->bytespersample;
+ }
+
+ readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + 1;
+
+ if (!isnotcompressed) {
+ /* compressed */
+ int16_t predictor_coef_table_a[32];
+ int predictor_coef_num_a;
+ int prediction_type_a;
+ int prediction_quantitization_a;
+ int ricemodifier_a;
+
+ int16_t predictor_coef_table_b[32];
+ int predictor_coef_num_b;
+ int prediction_type_b;
+ int prediction_quantitization_b;
+ int ricemodifier_b;
+
+ int i;
+
+ interlacing_shift = readbits(alac, 8);
+ interlacing_leftweight = readbits(alac, 8);
+
+ /******** channel 1 ***********/
+ prediction_type_a = readbits(alac, 4);
+ prediction_quantitization_a = readbits(alac, 4);
+
+ ricemodifier_a = readbits(alac, 3);
+ predictor_coef_num_a = readbits(alac, 5);
+
+ /* read the predictor table */
+ for (i = 0; i < predictor_coef_num_a; i++) {
+ predictor_coef_table_a[i] = (int16_t)readbits(alac, 16);
+ }
+
+ /******** channel 2 *********/
+ prediction_type_b = readbits(alac, 4);
+ prediction_quantitization_b = readbits(alac, 4);
+
+ ricemodifier_b = readbits(alac, 3);
+ predictor_coef_num_b = readbits(alac, 5);
+
+ /* read the predictor table */
+ for (i = 0; i < predictor_coef_num_b; i++) {
+ predictor_coef_table_b[i] = (int16_t)readbits(alac, 16);
+ }
+
+ /*********************/
+ if (wasted_bytes) {
+ /* see mono case */
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
+ }
+
+ /* channel 1 */
+ bastardized_rice_decompress(alac,
+ alac->predicterror_buffer_a,
+ outputsamples,
+ readsamplesize,
+ alac->setinfo_rice_initialhistory,
+ alac->setinfo_rice_kmodifier,
+ ricemodifier_a * alac->setinfo_rice_historymult / 4,
+ (1 << alac->setinfo_rice_kmodifier) - 1);
+
+ if (prediction_type_a == 0) {
+ /* adaptive fir */
+ predictor_decompress_fir_adapt(alac->predicterror_buffer_a,
+ alac->outputsamples_buffer_a,
+ outputsamples,
+ readsamplesize,
+ predictor_coef_table_a,
+ predictor_coef_num_a,
+ prediction_quantitization_a);
+ } else {
+ /* see mono case */
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_a);
+ }
+
+ /* channel 2 */
+ bastardized_rice_decompress(alac,
+ alac->predicterror_buffer_b,
+ outputsamples,
+ readsamplesize,
+ alac->setinfo_rice_initialhistory,
+ alac->setinfo_rice_kmodifier,
+ ricemodifier_b * alac->setinfo_rice_historymult / 4,
+ (1 << alac->setinfo_rice_kmodifier) - 1);
+
+ if (prediction_type_b == 0) {
+ /* adaptive fir */
+ predictor_decompress_fir_adapt(alac->predicterror_buffer_b,
+ alac->outputsamples_buffer_b,
+ outputsamples,
+ readsamplesize,
+ predictor_coef_table_b,
+ predictor_coef_num_b,
+ prediction_quantitization_b);
+ } else {
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type_b);
+ }
+ } else {
+ /* not compressed, easy case */
+ if (alac->setinfo_sample_size <= 16) {
+ int i;
+ for (i = 0; i < outputsamples; i++) {
+ int32_t audiobits_a, audiobits_b;
+
+ audiobits_a = readbits(alac, alac->setinfo_sample_size);
+ audiobits_b = readbits(alac, alac->setinfo_sample_size);
+
+ audiobits_a = SIGN_EXTENDED32(audiobits_a, alac->setinfo_sample_size);
+ audiobits_b = SIGN_EXTENDED32(audiobits_b, alac->setinfo_sample_size);
+
+ alac->outputsamples_buffer_a[i] = audiobits_a;
+ alac->outputsamples_buffer_b[i] = audiobits_b;
+ }
+ } else {
+ int i;
+ for (i = 0; i < outputsamples; i++) {
+ int32_t audiobits_a, audiobits_b;
+
+ audiobits_a = readbits(alac, 16);
+ audiobits_a = audiobits_a << 16;
+ audiobits_a = audiobits_a >> (32 - alac->setinfo_sample_size);
+ audiobits_a |= readbits(alac, alac->setinfo_sample_size - 16);
+
+ audiobits_b = readbits(alac, 16);
+ audiobits_b = audiobits_b << 16;
+ audiobits_b = audiobits_b >> (32 - alac->setinfo_sample_size);
+ audiobits_b |= readbits(alac, alac->setinfo_sample_size - 16);
+
+ alac->outputsamples_buffer_a[i] = audiobits_a;
+ alac->outputsamples_buffer_b[i] = audiobits_b;
+ }
+ }
+ /* wasted_bytes = 0; */
+ interlacing_shift = 0;
+ interlacing_leftweight = 0;
+ }
+
+ switch(alac->setinfo_sample_size) {
+ case 16: {
+ deinterlace_16(alac->outputsamples_buffer_a,
+ alac->outputsamples_buffer_b,
+ (int16_t*)outbuffer,
+ alac->numchannels,
+ outputsamples,
+ interlacing_shift,
+ interlacing_leftweight);
+ break;
+ }
+ case 20:
+ case 24:
+ case 32:
+ av_log(NULL, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
+ break;
+ default:
+ break;
+ }
+
+ break;
+ }
+ }
+
+av_log(NULL, AV_LOG_INFO, "buf size = %d, consumed %d\n",
+ input_buffer_size, alac->input_buffer_index);
+
+ /* avoid infinite loop: if decoder consumed 0 bytes; report all bytes
+ * consumed */
+// if (alac->input_buffer_index)
+// return alac->input_buffer_index;
+// else
+ return input_buffer_size;
+}
+
+static int alac_decode_init(AVCodecContext * avctx)
+{
+ ALACContext *s = avctx->priv_data;
+ s->avctx = avctx;
+ s->context_initialized = 0;
+
+ s->alac = av_malloc(sizeof(alac_file));
+
+ s->alac->samplesize = s->avctx->bits_per_sample;
+ s->alac->numchannels = s->avctx->channels;
+ s->alac->bytespersample = (s->alac->samplesize / 8) * s->alac->numchannels;
+
+ return 0;
+}
+
+static int alac_decode_frame(AVCodecContext *avctx,
+ void *data, int *data_size,
+ uint8_t *buf, int buf_size)
+{
+ ALACContext *s = avctx->priv_data;
+ int bytes_consumed = buf_size;
+
+ if (buf)
+ bytes_consumed = decode_frame(s, s->alac, buf, buf_size,
+ data, data_size);
+
+ return bytes_consumed;
+}
+
+static int alac_decode_close(AVCodecContext *avctx)
+{
+ ALACContext *s = avctx->priv_data;
+
+ av_free(s->alac->predicterror_buffer_a);
+ av_free(s->alac->predicterror_buffer_b);
+
+ av_free(s->alac->outputsamples_buffer_a);
+ av_free(s->alac->outputsamples_buffer_b);
+
+ return 0;
+}
+
+AVCodec alac_decoder = {
+ "alac",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_ALAC,
+ sizeof(ALACContext),
+ alac_decode_init,
+ NULL,
+ alac_decode_close,
+ alac_decode_frame,
+};
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 285cfcf..bd3707b 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -196,6 +196,7 @@ void avcodec_register_all(void)
register_avcodec(&qtrle_decoder);
register_avcodec(&flac_decoder);
register_avcodec(&shorten_decoder);
+ register_avcodec(&alac_decoder);
#endif /* CONFIG_DECODERS */
#ifdef AMR_NB
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index b4ab5b3..2c3780f 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -17,7 +17,7 @@ extern "C" {
#define FFMPEG_VERSION_INT 0x000409
#define FFMPEG_VERSION "0.4.9-pre1"
-#define LIBAVCODEC_BUILD 4744
+#define LIBAVCODEC_BUILD 4745
#define LIBAVCODEC_VERSION_INT FFMPEG_VERSION_INT
#define LIBAVCODEC_VERSION FFMPEG_VERSION
@@ -170,6 +170,7 @@ enum CodecID {
CODEC_ID_MPEG2TS= 0x20000, /* _FAKE_ codec to indicate a raw MPEG2 transport
stream (only used by libavformat) */
+ CODEC_ID_ALAC,
};
/* CODEC_ID_MP3LAME is absolete */
@@ -2011,6 +2012,7 @@ extern AVCodec xl_decoder;
extern AVCodec qpeg_decoder;
extern AVCodec shorten_decoder;
extern AVCodec loco_decoder;
+extern AVCodec alac_decoder;
/* pcm codecs */
#define PCM_CODEC(id, name) \
diff --git a/libavformat/mov.c b/libavformat/mov.c
index afa0a23..c0d9cec 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -142,6 +142,7 @@ static const CodecTag mov_audio_tags[] = {
{ CODEC_ID_AMR_NB, MKTAG('s', 'a', 'm', 'r') }, /* AMR-NB 3gp */
{ CODEC_ID_AMR_WB, MKTAG('s', 'a', 'w', 'b') }, /* AMR-WB 3gp */
{ CODEC_ID_AC3, MKTAG('m', 's', 0x20, 0x00) }, /* Dolby AC-3 */
+ { CODEC_ID_ALAC,MKTAG('a', 'l', 'a', 'c') }, /* Apple Lossless */
{ CODEC_ID_NONE, 0 },
};
@@ -1101,6 +1102,23 @@ static int mov_read_stsd(MOVContext *c, ByteIOContext *pb, MOV_atom_t atom)
st->codec.channels = (px[1] >> 3) & 15;
}
}
+ else if( st->codec.codec_tag == MKTAG( 'a', 'l', 'a', 'c' ))
+ {
+ /* Handle alac audio tag + special extradata */
+ get_be32(pb); /* version */
+ get_be32(pb);
+ st->codec.channels = get_be16(pb); /* channels */
+ st->codec.bits_per_sample = get_be16(pb); /* bits per sample */
+ get_be32(pb);
+ st->codec.sample_rate = get_be16(pb);
+ get_be16(pb);
+
+ /* fetch the 36-byte extradata needed for alac decoding */
+ st->codec.extradata_size = 36;
+ st->codec.extradata = (uint8_t*)
+ av_mallocz(st->codec.extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
+ get_buffer(pb, st->codec.extradata, st->codec.extradata_size);
+ }
else if(size>=(16+20))
{//16 bytes read, reading atleast 20 more
uint16_t version;
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