#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)
+/**
+ * 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 bs_code, sr_code, bps_code, i;
- int ch_mode, bps, blocksize, samplerate, channels;
- GetBitContext *gb = &s->gb;
+ int bs_code, sr_code, bps_code;
/* frame sync code */
- skip_bits(&s->gb, 16);
+ 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 */
- ch_mode = get_bits(gb, 4);
- if (ch_mode < FLAC_MAX_CHANNELS) {
- channels = ch_mode + 1;
- ch_mode = FLAC_CHMODE_INDEPENDENT;
- } else if (ch_mode <= FLAC_CHMODE_MID_SIDE) {
- channels = 2;
+ 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(s->avctx, AV_LOG_ERROR, "invalid channel mode: %d\n", ch_mode);
+ av_log(avctx, AV_LOG_ERROR, "invalid channel mode: %d\n", fi->ch_mode);
return -1;
}
/* bits per sample */
bps_code = get_bits(gb, 3);
if (bps_code == 3 || bps_code == 7) {
- av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n",
+ av_log(avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n",
bps_code);
return -1;
}
- bps = sample_size_table[bps_code];
+ 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;
}
/* blocksize */
if (bs_code == 0) {
- av_log(s->avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
+ av_log(avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
return -1;
} else if (bs_code == 6) {
- blocksize = get_bits(gb, 8) + 1;
+ fi->blocksize = get_bits(gb, 8) + 1;
} else if (bs_code == 7) {
- blocksize = get_bits(gb, 16) + 1;
+ fi->blocksize = get_bits(gb, 16) + 1;
} else {
- blocksize = ff_flac_blocksize_table[bs_code];
+ fi->blocksize = ff_flac_blocksize_table[bs_code];
}
/* sample rate */
if (sr_code < 12) {
- samplerate = ff_flac_sample_rate_table[sr_code];
+ fi->samplerate = ff_flac_sample_rate_table[sr_code];
} else if (sr_code == 12) {
- samplerate = get_bits(gb, 8) * 1000;
+ fi->samplerate = get_bits(gb, 8) * 1000;
} else if (sr_code == 13) {
- samplerate = get_bits(gb, 16);
+ fi->samplerate = get_bits(gb, 16);
} else if (sr_code == 14) {
- samplerate = get_bits(gb, 16) * 10;
+ fi->samplerate = get_bits(gb, 16) * 10;
} else {
- av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
+ av_log(avctx, AV_LOG_ERROR, "illegal sample rate code %d\n",
sr_code);
return -1;
}
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(s->avctx, AV_LOG_ERROR, "header crc mismatch\n");
+ av_log(avctx, AV_LOG_ERROR, "header crc mismatch\n");
+ return -1;
+ }
+
+ 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 (channels != s->channels) {
+ 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 = ch_mode;
+ s->ch_mode = fi.ch_mode;
- if (bps && bps != s->bps) {
+ if (fi.bps && fi.bps != s->bps) {
av_log(s->avctx, AV_LOG_ERROR, "switching bps mid-stream is not "
"supported\n");
return -1;
s->is32 = 0;
}
- if (blocksize > s->max_blocksize) {
- av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", 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 = blocksize;
+ s->blocksize = fi.blocksize;
- if (samplerate == 0) {
- samplerate = s->samplerate;
- } else if (samplerate != s->samplerate) {
+ 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, samplerate);
+ s->samplerate, fi.samplerate);
}
- s->samplerate = s->avctx->sample_rate = 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;
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