#include "libavutil/crc.h"
#include "avcodec.h"
#include "internal.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "bytestream.h"
#include "golomb.h"
#include "flac.h"
/* initialize based on the demuxer-supplied streamdata header */
ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
+ if (s->bps > 16)
+ avctx->sample_fmt = SAMPLE_FMT_S32;
+ else
+ avctx->sample_fmt = SAMPLE_FMT_S16;
allocate_buffers(s);
s->got_streaminfo = 1;
avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
avctx->bits_per_raw_sample = s->bps;
- if (s->bps > 16)
- avctx->sample_fmt = SAMPLE_FMT_S32;
- else
- avctx->sample_fmt = SAMPLE_FMT_S16;
s->samples = get_bits_long(&gb, 32) << 4;
s->samples |= get_bits(&gb, 4);
{
int i, j;
int coeff_prec, qlevel;
- int coeffs[pred_order];
+ int coeffs[32];
int32_t *decoded = s->decoded[channel];
/* warm up samples */
s->curr_bps -= wasted;
}
if (s->curr_bps > 32) {
- ff_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
+ av_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
return -1;
}
return 0;
}
-static int decode_frame(FLACContext *s, int alloc_data_size)
+/**
+ * Validate and decode a frame header.
+ * @param avctx AVCodecContext to use as av_log() context
+ * @param gb GetBitContext from which to read frame header
+ * @param[out] fi frame information
+ * @return non-zero on error, 0 if ok
+ */
+static int decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
+ FLACFrameInfo *fi)
{
- int blocksize_code, sample_rate_code, sample_size_code, i, crc8;
- int ch_mode, bps, blocksize, samplerate;
- GetBitContext *gb = &s->gb;
-
- blocksize_code = get_bits(gb, 4);
-
- sample_rate_code = get_bits(gb, 4);
-
- ch_mode = get_bits(gb, 4); /* channel assignment */
- if (ch_mode < FLAC_MAX_CHANNELS && s->channels == ch_mode+1) {
- ch_mode = FLAC_CHMODE_INDEPENDENT;
- } else if (ch_mode > FLAC_CHMODE_MID_SIDE || s->channels != 2) {
- av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n",
- ch_mode, s->channels);
+ int bs_code, sr_code, bps_code;
+
+ /* frame sync code */
+ skip_bits(gb, 16);
+
+ /* block size and sample rate codes */
+ bs_code = get_bits(gb, 4);
+ sr_code = get_bits(gb, 4);
+
+ /* channels and decorrelation */
+ fi->ch_mode = get_bits(gb, 4);
+ if (fi->ch_mode < FLAC_MAX_CHANNELS) {
+ fi->channels = fi->ch_mode + 1;
+ fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
+ } else if (fi->ch_mode <= FLAC_CHMODE_MID_SIDE) {
+ fi->channels = 2;
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "invalid channel mode: %d\n", fi->ch_mode);
return -1;
}
- sample_size_code = get_bits(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);
+ /* bits per sample */
+ bps_code = get_bits(gb, 3);
+ if (bps_code == 3 || bps_code == 7) {
+ av_log(avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n",
+ bps_code);
return -1;
}
- if (bps > 16) {
- s->avctx->sample_fmt = SAMPLE_FMT_S32;
- s->sample_shift = 32 - bps;
- s->is32 = 1;
- } else {
- s->avctx->sample_fmt = SAMPLE_FMT_S16;
- s->sample_shift = 16 - bps;
- s->is32 = 0;
- }
- s->bps = s->avctx->bits_per_raw_sample = bps;
+ fi->bps = sample_size_table[bps_code];
+ /* reserved bit */
if (get_bits1(gb)) {
- av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
+ av_log(avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
return -1;
}
+ /* sample or frame count */
if (get_utf8(gb) < 0) {
- av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
+ av_log(avctx, AV_LOG_ERROR, "utf8 fscked\n");
return -1;
}
- if (blocksize_code == 0) {
- av_log(s->avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
+ /* blocksize */
+ if (bs_code == 0) {
+ av_log(avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
return -1;
- } else if (blocksize_code == 6)
- blocksize = get_bits(gb, 8)+1;
- else if (blocksize_code == 7)
- blocksize = get_bits(gb, 16)+1;
- else
- blocksize = ff_flac_blocksize_table[blocksize_code];
+ } else if (bs_code == 6) {
+ fi->blocksize = get_bits(gb, 8) + 1;
+ } else if (bs_code == 7) {
+ fi->blocksize = get_bits(gb, 16) + 1;
+ } else {
+ fi->blocksize = ff_flac_blocksize_table[bs_code];
+ }
+
+ /* sample rate */
+ if (sr_code < 12) {
+ fi->samplerate = ff_flac_sample_rate_table[sr_code];
+ } else if (sr_code == 12) {
+ fi->samplerate = get_bits(gb, 8) * 1000;
+ } else if (sr_code == 13) {
+ fi->samplerate = get_bits(gb, 16);
+ } else if (sr_code == 14) {
+ fi->samplerate = get_bits(gb, 16) * 10;
+ } else {
+ av_log(avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
+ sr_code);
+ return -1;
+ }
- if (blocksize > s->max_blocksize) {
- av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize,
- s->max_blocksize);
+ /* header CRC-8 check */
+ skip_bits(gb, 8);
+ if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
+ get_bits_count(gb)/8)) {
+ av_log(avctx, AV_LOG_ERROR, "header crc mismatch\n");
return -1;
}
- if (blocksize * s->channels * (s->is32 ? 4 : 2) > alloc_data_size)
+ return 0;
+}
+
+static int decode_frame(FLACContext *s)
+{
+ int i;
+ GetBitContext *gb = &s->gb;
+ FLACFrameInfo fi;
+
+ if (decode_frame_header(s->avctx, gb, &fi)) {
+ av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n");
return -1;
+ }
- if (sample_rate_code == 0)
- samplerate= s->samplerate;
- else if (sample_rate_code < 12)
- samplerate = ff_flac_sample_rate_table[sample_rate_code];
- else if (sample_rate_code == 12)
- samplerate = get_bits(gb, 8) * 1000;
- else if (sample_rate_code == 13)
- samplerate = get_bits(gb, 16);
- else if (sample_rate_code == 14)
- samplerate = get_bits(gb, 16) * 10;
- else {
- av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
- sample_rate_code);
+ if (fi.channels != s->channels) {
+ av_log(s->avctx, AV_LOG_ERROR, "switching channel layout mid-stream "
+ "is not supported\n");
return -1;
}
+ s->ch_mode = fi.ch_mode;
- skip_bits(gb, 8);
- crc8 = av_crc(av_crc_get_table(AV_CRC_8_ATM), 0,
- gb->buffer, get_bits_count(gb)/8);
- if (crc8) {
- av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
+ if (fi.bps && fi.bps != s->bps) {
+ av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
+ "supported\n");
+ return -1;
+ }
+ if (s->bps > 16) {
+ s->avctx->sample_fmt = SAMPLE_FMT_S32;
+ s->sample_shift = 32 - s->bps;
+ s->is32 = 1;
+ } else {
+ s->avctx->sample_fmt = SAMPLE_FMT_S16;
+ s->sample_shift = 16 - s->bps;
+ s->is32 = 0;
+ }
+
+ 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;
- s->blocksize = blocksize;
- s->samplerate = samplerate;
- s->bps = bps;
- s->ch_mode = ch_mode;
+ if (fi.samplerate == 0) {
+ fi.samplerate = s->samplerate;
+ } else if (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;
// dump_headers(s->avctx, (FLACStreaminfo *)s);
static int flac_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
- const uint8_t *buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
FLACContext *s = avctx->priv_data;
int i, j = 0, input_buf_size = 0, bytes_read = 0;
int16_t *samples_16 = data;
int32_t *samples_32 = data;
int alloc_data_size= *data_size;
+ int output_size;
*data_size=0;
/* decode frame */
init_get_bits(&s->gb, buf, buf_size*8);
- skip_bits(&s->gb, 16);
- if (decode_frame(s, alloc_data_size) < 0) {
+ if (decode_frame(s) < 0) {
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
s->bitstream_size=0;
s->bitstream_index=0;
return -1;
}
- *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
bytes_read = (get_bits_count(&s->gb)+7)/8;
+ /* check if allocated data size is large enough for output */
+ output_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
+ if (output_size > alloc_data_size) {
+ av_log(s->avctx, AV_LOG_ERROR, "output data size is larger than "
+ "allocated data size\n");
+ goto end;
+ }
+ *data_size = output_size;
+
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
for (i = 0; i < s->blocksize; i++) {\
NULL,
flac_decode_close,
flac_decode_frame,
- CODEC_CAP_DELAY,
+ CODEC_CAP_DELAY | CODEC_CAP_SUBFRAMES, /* FIXME: add a FLAC parser so that
+ we will not need to use either
+ of these capabilities */
.flush= flac_flush,
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
};
projects / frescor / ffmpeg.git / blobdiff