* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
- * 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 flacdec.c
+ * @file
* FLAC (Free Lossless Audio Codec) decoder
* @author Alex Beregszaszi
- *
- * For more information on the FLAC format, visit:
- * http://flac.sourceforge.net/
+ * @see http://flac.sourceforge.net/
*
* This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
* through, starting from the initial 'fLaC' signature; or by passing the
#include <limits.h>
-#define ALT_BITSTREAM_READER
+#include "libavutil/audioconvert.h"
#include "libavutil/crc.h"
#include "avcodec.h"
-#include "bitstream.h"
+#include "internal.h"
+#include "get_bits.h"
+#include "bytestream.h"
#include "golomb.h"
#include "flac.h"
+#include "flacdata.h"
#undef NDEBUG
#include <assert.h>
-#define MAX_CHANNELS 8
-#define MAX_BLOCKSIZE 65535
-#define FLAC_STREAMINFO_SIZE 34
-
-enum decorrelation_type {
- INDEPENDENT,
- LEFT_SIDE,
- RIGHT_SIDE,
- MID_SIDE,
-};
-
typedef struct FLACContext {
FLACSTREAMINFO
- AVCodecContext *avctx;
- GetBitContext gb;
+ AVCodecContext *avctx; ///< parent AVCodecContext
+ AVFrame frame;
+ GetBitContext gb; ///< GetBitContext initialized to start at the current frame
- int blocksize/*, last_blocksize*/;
- int curr_bps;
- enum decorrelation_type decorrelation;
+ int blocksize; ///< number of samples in the current frame
+ int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits
+ int sample_shift; ///< shift required to make output samples 16-bit or 32-bit
+ int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit
+ int ch_mode; ///< channel decorrelation type in the current frame
+ int got_streaminfo; ///< indicates if the STREAMINFO has been read
- int32_t *decoded[MAX_CHANNELS];
- uint8_t *bitstream;
- unsigned int bitstream_size;
- unsigned int bitstream_index;
- unsigned int allocated_bitstream_size;
+ int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
} FLACContext;
-#define METADATA_TYPE_STREAMINFO 0
-
-static const int sample_rate_table[] =
-{ 0,
- 88200, 176400, 192000,
- 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
- 0, 0, 0, 0 };
-
-static const int sample_size_table[] =
-{ 0, 8, 12, 0, 16, 20, 24, 0 };
-
-static const int blocksize_table[] = {
- 0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
-256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
+static const int64_t flac_channel_layouts[6] = {
+ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_QUAD,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_5POINT1
};
-static int64_t get_utf8(GetBitContext *gb){
- int64_t val;
- GET_UTF8(val, get_bits(gb, 8), return -1;)
- return val;
-}
-
static void allocate_buffers(FLACContext *s);
-static int metadata_parse(FLACContext *s);
-static av_cold int flac_decode_init(AVCodecContext * avctx)
+int avpriv_flac_is_extradata_valid(AVCodecContext *avctx,
+ enum FLACExtradataFormat *format,
+ uint8_t **streaminfo_start)
{
+ if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
+ return 0;
+ }
+ if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
+ /* extradata contains STREAMINFO only */
+ if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
+ av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
+ FLAC_STREAMINFO_SIZE-avctx->extradata_size);
+ }
+ *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
+ *streaminfo_start = avctx->extradata;
+ } else {
+ if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
+ av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
+ return 0;
+ }
+ *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
+ *streaminfo_start = &avctx->extradata[8];
+ }
+ return 1;
+}
+
+static av_cold int flac_decode_init(AVCodecContext *avctx)
+{
+ enum FLACExtradataFormat format;
+ uint8_t *streaminfo;
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
- if (avctx->extradata_size > 4) {
- /* initialize based on the demuxer-supplied streamdata header */
- if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
- ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata);
- allocate_buffers(s);
- } else {
- init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
- metadata_parse(s);
- }
- }
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+
+ /* for now, the raw FLAC header is allowed to be passed to the decoder as
+ frame data instead of extradata. */
+ if (!avctx->extradata)
+ return 0;
+
+ if (!avpriv_flac_is_extradata_valid(avctx, &format, &streaminfo))
+ return -1;
+
+ /* initialize based on the demuxer-supplied streamdata header */
+ avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
+ if (s->bps > 16)
+ avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ else
+ avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ allocate_buffers(s);
+ s->got_streaminfo = 1;
+
+ avcodec_get_frame_defaults(&s->frame);
+ avctx->coded_frame = &s->frame;
+
+ if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))
+ avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];
- avctx->sample_fmt = SAMPLE_FMT_S16;
return 0;
}
static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
{
- av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize);
+ av_log(avctx, AV_LOG_DEBUG, " Max Blocksize: %d\n", s->max_blocksize);
av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize);
av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
av_log(avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps);
}
-static void allocate_buffers(FLACContext *s){
+static void allocate_buffers(FLACContext *s)
+{
int i;
assert(s->max_blocksize);
- if(s->max_framesize == 0 && s->max_blocksize){
- s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead
- }
-
- for (i = 0; i < s->channels; i++)
- {
- s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize);
+ for (i = 0; i < s->channels; i++) {
+ s->decoded[i] = av_realloc(s->decoded[i],
+ sizeof(int32_t)*s->max_blocksize);
}
-
- if(s->allocated_bitstream_size < s->max_framesize)
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
}
-void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
+void avpriv_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
const uint8_t *buffer)
{
GetBitContext gb;
init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
- /* mandatory streaminfo */
- s->min_blocksize = get_bits(&gb, 16);
+ skip_bits(&gb, 16); /* skip min blocksize */
s->max_blocksize = get_bits(&gb, 16);
+ if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
+ av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
+ s->max_blocksize);
+ s->max_blocksize = 16;
+ }
skip_bits(&gb, 24); /* skip min frame size */
s->max_framesize = get_bits_long(&gb, 24);
avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
+ avctx->bits_per_raw_sample = s->bps;
- skip_bits(&gb, 36); /* total num of samples */
+ s->samples = get_bits_long(&gb, 32) << 4;
+ s->samples |= get_bits(&gb, 4);
- skip_bits(&gb, 64); /* md5 sum */
- skip_bits(&gb, 64); /* md5 sum */
+ skip_bits_long(&gb, 64); /* md5 sum */
+ skip_bits_long(&gb, 64); /* md5 sum */
dump_headers(avctx, s);
}
+void avpriv_flac_parse_block_header(const uint8_t *block_header,
+ int *last, int *type, int *size)
+{
+ int tmp = bytestream_get_byte(&block_header);
+ if (last)
+ *last = tmp & 0x80;
+ if (type)
+ *type = tmp & 0x7F;
+ if (size)
+ *size = bytestream_get_be24(&block_header);
+}
+
/**
- * Parse a list of metadata blocks. This list of blocks must begin with
- * the fLaC marker.
- * @param s the flac decoding context containing the gb bit reader used to
- * parse metadata
- * @return 1 if some metadata was read, 0 if no fLaC marker was found
+ * Parse the STREAMINFO from an inline header.
+ * @param s the flac decoding context
+ * @param buf input buffer, starting with the "fLaC" marker
+ * @param buf_size buffer size
+ * @return non-zero if metadata is invalid
*/
-static int metadata_parse(FLACContext *s)
+static int parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size)
{
- int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0;
- int initial_pos= get_bits_count(&s->gb);
-
- if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
- skip_bits(&s->gb, 32);
+ int metadata_type, metadata_size;
- do {
- metadata_last = get_bits1(&s->gb);
- metadata_type = get_bits(&s->gb, 7);
- metadata_size = get_bits_long(&s->gb, 24);
-
- if(get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits){
- skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb));
- break;
- }
+ if (buf_size < FLAC_STREAMINFO_SIZE+8) {
+ /* need more data */
+ return 0;
+ }
+ avpriv_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
+ if (metadata_type != FLAC_METADATA_TYPE_STREAMINFO ||
+ metadata_size != FLAC_STREAMINFO_SIZE) {
+ return AVERROR_INVALIDDATA;
+ }
+ avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
+ allocate_buffers(s);
+ s->got_streaminfo = 1;
- if (metadata_size) {
- switch (metadata_type) {
- case METADATA_TYPE_STREAMINFO:
- ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8);
- streaminfo_updated = 1;
+ return 0;
+}
- default:
- for (i=0; i<metadata_size; i++)
- skip_bits(&s->gb, 8);
- }
- }
- } while (!metadata_last);
+/**
+ * Determine the size of an inline header.
+ * @param buf input buffer, starting with the "fLaC" marker
+ * @param buf_size buffer size
+ * @return number of bytes in the header, or 0 if more data is needed
+ */
+static int get_metadata_size(const uint8_t *buf, int buf_size)
+{
+ int metadata_last, metadata_size;
+ const uint8_t *buf_end = buf + buf_size;
+
+ buf += 4;
+ do {
+ if (buf_end - buf < 4)
+ return 0;
+ avpriv_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
+ buf += 4;
+ if (buf_end - buf < metadata_size) {
+ /* need more data in order to read the complete header */
+ return 0;
+ }
+ buf += metadata_size;
+ } while (!metadata_last);
- if (streaminfo_updated)
- allocate_buffers(s);
- return 1;
- }
- return 0;
+ return buf_size - (buf_end - buf);
}
static int decode_residuals(FLACContext *s, int channel, int pred_order)
int sample = 0, samples;
method_type = get_bits(&s->gb, 2);
- if (method_type > 1){
- av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n", method_type);
+ if (method_type > 1) {
+ av_log(s->avctx, AV_LOG_ERROR, "illegal residual coding method %d\n",
+ method_type);
return -1;
}
samples= s->blocksize >> rice_order;
if (pred_order > samples) {
- av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples);
+ av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n",
+ pred_order, samples);
return -1;
}
sample=
i= pred_order;
- for (partition = 0; partition < (1 << rice_order); partition++)
- {
+ for (partition = 0; partition < (1 << rice_order); partition++) {
tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
- if (tmp == (method_type == 0 ? 15 : 31))
- {
+ if (tmp == (method_type == 0 ? 15 : 31)) {
tmp = get_bits(&s->gb, 5);
for (; i < samples; i++, sample++)
- s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
- }
- else
- {
- for (; i < samples; i++, sample++){
+ s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp);
+ } else {
+ for (; i < samples; i++, sample++) {
s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
}
}
int a, b, c, d, i;
/* warm up samples */
- for (i = 0; i < pred_order; i++)
- {
- decoded[i] = get_sbits(&s->gb, s->curr_bps);
+ for (i = 0; i < pred_order; i++) {
+ decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
}
if (decode_residuals(s, channel, pred_order) < 0)
return -1;
- if(pred_order > 0)
+ if (pred_order > 0)
a = decoded[pred_order-1];
- if(pred_order > 1)
+ if (pred_order > 1)
b = a - decoded[pred_order-2];
- if(pred_order > 2)
+ if (pred_order > 2)
c = b - decoded[pred_order-2] + decoded[pred_order-3];
- if(pred_order > 3)
+ if (pred_order > 3)
d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4];
- switch(pred_order)
- {
- case 0:
- break;
- case 1:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += decoded[i];
- break;
- case 2:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += decoded[i];
- break;
- case 3:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += c += decoded[i];
- break;
- case 4:
- for (i = pred_order; i < blocksize; i++)
- decoded[i] = a += b += c += d += decoded[i];
- break;
- default:
- av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
- return -1;
+ switch (pred_order) {
+ case 0:
+ break;
+ case 1:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = a += decoded[i];
+ break;
+ case 2:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = a += b += decoded[i];
+ break;
+ case 3:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = a += b += c += decoded[i];
+ break;
+ case 4:
+ for (i = pred_order; i < blocksize; i++)
+ decoded[i] = a += b += c += d += decoded[i];
+ break;
+ default:
+ av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
+ return -1;
}
return 0;
{
int i, j;
int coeff_prec, qlevel;
- int coeffs[pred_order];
+ int coeffs[32];
int32_t *decoded = s->decoded[channel];
/* warm up samples */
- for (i = 0; i < pred_order; i++)
- {
- decoded[i] = get_sbits(&s->gb, s->curr_bps);
+ for (i = 0; i < pred_order; i++) {
+ decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
}
coeff_prec = get_bits(&s->gb, 4) + 1;
- if (coeff_prec == 16)
- {
+ if (coeff_prec == 16) {
av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
return -1;
}
qlevel = get_sbits(&s->gb, 5);
- if(qlevel < 0){
- av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n", qlevel);
+ if (qlevel < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "qlevel %d not supported, maybe buggy stream\n",
+ qlevel);
return -1;
}
- for (i = 0; i < pred_order; i++)
- {
+ for (i = 0; i < pred_order; i++) {
coeffs[i] = get_sbits(&s->gb, coeff_prec);
}
if (s->bps > 16) {
int64_t sum;
- for (i = pred_order; i < s->blocksize; i++)
- {
+ for (i = pred_order; i < s->blocksize; i++) {
sum = 0;
for (j = 0; j < pred_order; j++)
sum += (int64_t)coeffs[j] * decoded[i-j-1];
decoded[i] += sum >> qlevel;
}
} else {
- for (i = pred_order; i < s->blocksize-1; i += 2)
- {
+ for (i = pred_order; i < s->blocksize-1; i += 2) {
int c;
int d = decoded[i-pred_order];
int s0 = 0, s1 = 0;
- for (j = pred_order-1; j > 0; j--)
- {
+ for (j = pred_order-1; j > 0; j--) {
c = coeffs[j];
s0 += c*d;
d = decoded[i-j];
s1 += c*d;
decoded[i+1] += s1 >> qlevel;
}
- if (i < s->blocksize)
- {
+ if (i < s->blocksize) {
int sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * decoded[i-j-1];
int i, tmp;
s->curr_bps = s->bps;
- if(channel == 0){
- if(s->decorrelation == RIGHT_SIDE)
+ if (channel == 0) {
+ if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
s->curr_bps++;
- }else{
- if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
+ } else {
+ if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
s->curr_bps++;
}
- if (get_bits1(&s->gb))
- {
+ if (get_bits1(&s->gb)) {
av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
return -1;
}
type = get_bits(&s->gb, 6);
- if (get_bits1(&s->gb))
- {
+ if (get_bits1(&s->gb)) {
+ int left = get_bits_left(&s->gb);
wasted = 1;
+ if ( left < 0 ||
+ (left < s->curr_bps && !show_bits_long(&s->gb, left)) ||
+ !show_bits_long(&s->gb, s->curr_bps)) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid number of wasted bits > available bits (%d) - left=%d\n",
+ s->curr_bps, left);
+ return AVERROR_INVALIDDATA;
+ }
while (!get_bits1(&s->gb))
wasted++;
s->curr_bps -= wasted;
}
+ if (s->curr_bps > 32) {
+ av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
+ return -1;
+ }
//FIXME use av_log2 for types
- if (type == 0)
- {
- tmp = get_sbits(&s->gb, s->curr_bps);
+ if (type == 0) {
+ tmp = get_sbits_long(&s->gb, s->curr_bps);
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] = tmp;
- }
- else if (type == 1)
- {
+ } else if (type == 1) {
for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
- }
- else if ((type >= 8) && (type <= 12))
- {
+ s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps);
+ } else if ((type >= 8) && (type <= 12)) {
if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
return -1;
- }
- else if (type >= 32)
- {
+ } else if (type >= 32) {
if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
return -1;
- }
- else
- {
+ } else {
av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
return -1;
}
- if (wasted)
- {
+ if (wasted) {
int i;
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] <<= wasted;
return 0;
}
-static int decode_frame(FLACContext *s, int alloc_data_size)
+static int decode_frame(FLACContext *s)
{
- int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
- int decorrelation, bps, blocksize, samplerate;
-
- blocksize_code = get_bits(&s->gb, 4);
-
- sample_rate_code = get_bits(&s->gb, 4);
+ int i;
+ GetBitContext *gb = &s->gb;
+ FLACFrameInfo fi;
- assignment = get_bits(&s->gb, 4); /* channel assignment */
- if (assignment < 8 && s->channels == assignment+1)
- decorrelation = INDEPENDENT;
- else if (assignment >=8 && assignment < 11 && s->channels == 2)
- decorrelation = LEFT_SIDE + assignment - 8;
- else
- {
- av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
+ if (ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
return -1;
}
- sample_size_code = get_bits(&s->gb, 3);
- if(sample_size_code == 0)
- bps= s->bps;
- else if((sample_size_code != 3) && (sample_size_code != 7))
- bps = sample_size_table[sample_size_code];
- else
- {
- av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
+ if (s->channels && fi.channels != s->channels) {
+ av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "
+ "is not supported\n");
return -1;
}
+ s->channels = s->avctx->channels = fi.channels;
+ s->ch_mode = fi.ch_mode;
- if (get_bits1(&s->gb))
- {
- av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
+ if (!s->bps && !fi.bps) {
+ av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n");
return -1;
}
-
- if(get_utf8(&s->gb) < 0){
- av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
+ if (!fi.bps) {
+ fi.bps = s->bps;
+ } else if (s->bps && fi.bps != s->bps) {
+ av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
+ "supported\n");
return -1;
}
+ s->bps = s->avctx->bits_per_raw_sample = fi.bps;
- if (blocksize_code == 0)
- blocksize = s->min_blocksize;
- else if (blocksize_code == 6)
- blocksize = get_bits(&s->gb, 8)+1;
- else if (blocksize_code == 7)
- blocksize = get_bits(&s->gb, 16)+1;
- else
- blocksize = blocksize_table[blocksize_code];
-
- if(blocksize > s->max_blocksize){
- av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize);
- return -1;
+ if (s->bps > 16) {
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ s->sample_shift = 32 - s->bps;
+ s->is32 = 1;
+ } else {
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->sample_shift = 16 - s->bps;
+ s->is32 = 0;
}
- if(blocksize * s->channels * sizeof(int16_t) > alloc_data_size)
- return -1;
-
- if (sample_rate_code == 0){
- samplerate= s->samplerate;
- }else if (sample_rate_code < 12)
- samplerate = sample_rate_table[sample_rate_code];
- else if (sample_rate_code == 12)
- samplerate = get_bits(&s->gb, 8) * 1000;
- else if (sample_rate_code == 13)
- samplerate = get_bits(&s->gb, 16);
- else if (sample_rate_code == 14)
- samplerate = get_bits(&s->gb, 16) * 10;
- else{
- av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code);
+ if (!s->max_blocksize)
+ s->max_blocksize = FLAC_MAX_BLOCKSIZE;
+ if (fi.blocksize > s->max_blocksize) {
+ av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", fi.blocksize,
+ s->max_blocksize);
return -1;
}
+ s->blocksize = fi.blocksize;
- skip_bits(&s->gb, 8);
- crc8 = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0,
- s->gb.buffer, get_bits_count(&s->gb)/8);
- if(crc8){
- av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
+ if (!s->samplerate && !fi.samplerate) {
+ av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO"
+ " or frame header\n");
return -1;
}
+ if (fi.samplerate == 0) {
+ fi.samplerate = s->samplerate;
+ } else if (s->samplerate && fi.samplerate != s->samplerate) {
+ av_log(s->avctx, AV_LOG_WARNING, "sample rate changed from %d to %d\n",
+ s->samplerate, fi.samplerate);
+ }
+ s->samplerate = s->avctx->sample_rate = fi.samplerate;
- s->blocksize = blocksize;
- s->samplerate = samplerate;
- s->bps = bps;
- s->decorrelation= decorrelation;
+ if (!s->got_streaminfo) {
+ allocate_buffers(s);
+ s->got_streaminfo = 1;
+ dump_headers(s->avctx, (FLACStreaminfo *)s);
+ }
// dump_headers(s->avctx, (FLACStreaminfo *)s);
/* subframes */
- for (i = 0; i < s->channels; i++)
- {
+ for (i = 0; i < s->channels; i++) {
if (decode_subframe(s, i) < 0)
return -1;
}
- align_get_bits(&s->gb);
+ align_get_bits(gb);
/* frame footer */
- skip_bits(&s->gb, 16); /* data crc */
+ skip_bits(gb, 16); /* data crc */
return 0;
}
-static int flac_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- const uint8_t *buf, int buf_size)
+static int flac_decode_frame(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
- int tmp = 0, i, j = 0, input_buf_size = 0;
- int16_t *samples = data;
- int alloc_data_size= *data_size;
+ int i, j = 0, bytes_read = 0;
+ int16_t *samples_16;
+ int32_t *samples_32;
+ int ret;
- *data_size=0;
+ *got_frame_ptr = 0;
- if(s->max_framesize == 0){
- s->max_framesize= FFMAX(4, buf_size); // should hopefully be enough for the first header
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
+ if (s->max_framesize == 0) {
+ s->max_framesize =
+ ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
+ FLAC_MAX_CHANNELS, 32);
}
- if(1 && s->max_framesize){//FIXME truncated
- if(s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
- buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
- input_buf_size= buf_size;
-
- if(s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
- return -1;
-
- if(s->allocated_bitstream_size < s->bitstream_size + buf_size)
- s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
+ /* check that there is at least the smallest decodable amount of data.
+ this amount corresponds to the smallest valid FLAC frame possible.
+ FF F8 69 02 00 00 9A 00 00 34 46 */
+ if (buf_size < FLAC_MIN_FRAME_SIZE)
+ return buf_size;
- if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
- memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
- s->bitstream_index=0;
- }
- memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
- buf= &s->bitstream[s->bitstream_index];
- buf_size += s->bitstream_size;
- s->bitstream_size= buf_size;
-
- if(buf_size < s->max_framesize && input_buf_size){
- return input_buf_size;
- }
+ /* check for inline header */
+ if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
+ if (!s->got_streaminfo && parse_streaminfo(s, buf, buf_size)) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
+ return -1;
+ }
+ return get_metadata_size(buf, buf_size);
}
+ /* decode frame */
init_get_bits(&s->gb, buf, buf_size*8);
+ if (decode_frame(s) < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
+ return -1;
+ }
+ bytes_read = (get_bits_count(&s->gb)+7)/8;
- if(metadata_parse(s))
- goto end;
-
- tmp = show_bits(&s->gb, 16);
- if((tmp & 0xFFFE) != 0xFFF8){
- av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
- while(get_bits_count(&s->gb)/8+2 < buf_size && (show_bits(&s->gb, 16) & 0xFFFE) != 0xFFF8)
- skip_bits(&s->gb, 8);
- goto end; // we may not have enough bits left to decode a frame, so try next time
- }
- skip_bits(&s->gb, 16);
- if (decode_frame(s, alloc_data_size) < 0){
- av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
- s->bitstream_size=0;
- s->bitstream_index=0;
- return -1;
- }
+ /* get output buffer */
+ s->frame.nb_samples = s->blocksize;
+ if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
+ }
+ samples_16 = (int16_t *)s->frame.data[0];
+ samples_32 = (int32_t *)s->frame.data[0];
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
- for (i = 0; i < s->blocksize; i++)\
- {\
+ for (i = 0; i < s->blocksize; i++) {\
int a= s->decoded[0][i];\
int b= s->decoded[1][i];\
- *samples++ = ((left) << (24 - s->bps)) >> 8;\
- *samples++ = ((right) << (24 - s->bps)) >> 8;\
+ if (s->is32) {\
+ *samples_32++ = (left) << s->sample_shift;\
+ *samples_32++ = (right) << s->sample_shift;\
+ } else {\
+ *samples_16++ = (left) << s->sample_shift;\
+ *samples_16++ = (right) << s->sample_shift;\
+ }\
}\
break;
- switch(s->decorrelation)
- {
- case INDEPENDENT:
- for (j = 0; j < s->blocksize; j++)
- {
- for (i = 0; i < s->channels; i++)
- *samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8;
+ switch (s->ch_mode) {
+ case FLAC_CHMODE_INDEPENDENT:
+ for (j = 0; j < s->blocksize; j++) {
+ for (i = 0; i < s->channels; i++) {
+ if (s->is32)
+ *samples_32++ = s->decoded[i][j] << s->sample_shift;
+ else
+ *samples_16++ = s->decoded[i][j] << s->sample_shift;
}
- break;
- case LEFT_SIDE:
- DECORRELATE(a,a-b)
- case RIGHT_SIDE:
- DECORRELATE(a+b,b)
- case MID_SIDE:
- DECORRELATE( (a-=b>>1) + b, a)
- }
-
- *data_size = (int8_t *)samples - (int8_t *)data;
-
-end:
- i= (get_bits_count(&s->gb)+7)/8;
- if(i > buf_size){
- av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
- s->bitstream_size=0;
- s->bitstream_index=0;
+ }
+ break;
+ case FLAC_CHMODE_LEFT_SIDE:
+ DECORRELATE(a,a-b)
+ case FLAC_CHMODE_RIGHT_SIDE:
+ DECORRELATE(a+b,b)
+ case FLAC_CHMODE_MID_SIDE:
+ DECORRELATE( (a-=b>>1) + b, a)
+ }
+
+ if (bytes_read > buf_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
return -1;
}
+ if (bytes_read < buf_size) {
+ av_log(s->avctx, AV_LOG_DEBUG, "underread: %d orig size: %d\n",
+ buf_size - bytes_read, buf_size);
+ }
+
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
- if(s->bitstream_size){
- s->bitstream_index += i;
- s->bitstream_size -= i;
- return input_buf_size;
- }else
- return i;
+ return bytes_read;
}
static av_cold int flac_decode_close(AVCodecContext *avctx)
FLACContext *s = avctx->priv_data;
int i;
- for (i = 0; i < s->channels; i++)
- {
+ for (i = 0; i < s->channels; i++) {
av_freep(&s->decoded[i]);
}
- av_freep(&s->bitstream);
return 0;
}
-static void flac_flush(AVCodecContext *avctx){
- FLACContext *s = avctx->priv_data;
-
- s->bitstream_size=
- s->bitstream_index= 0;
-}
-
-AVCodec flac_decoder = {
- "flac",
- CODEC_TYPE_AUDIO,
- CODEC_ID_FLAC,
- sizeof(FLACContext),
- flac_decode_init,
- NULL,
- flac_decode_close,
- flac_decode_frame,
- CODEC_CAP_DELAY,
- .flush= flac_flush,
- .long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
+AVCodec ff_flac_decoder = {
+ .name = "flac",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_FLAC,
+ .priv_data_size = sizeof(FLACContext),
+ .init = flac_decode_init,
+ .close = flac_decode_close,
+ .decode = flac_decode_frame,
+ .capabilities = CODEC_CAP_DR1,
+ .long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};
projects / ffmpeg / blobdiff