* 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 libavcodec/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>
+#include "libavutil/audioconvert.h"
#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"
+#include "flacdata.h"
+#include "flacdsp.h"
#undef NDEBUG
#include <assert.h>
-#define MAX_CHANNELS 8
-#define MAX_BLOCKSIZE 65535
-
-enum decorrelation_type {
- INDEPENDENT,
- LEFT_SIDE,
- RIGHT_SIDE,
- MID_SIDE,
-};
-
typedef struct FLACContext {
FLACSTREAMINFO
AVCodecContext *avctx; ///< parent AVCodecContext
+ AVFrame frame;
GetBitContext gb; ///< GetBitContext initialized to start at the current frame
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
- enum decorrelation_type decorrelation; ///< channel decorrelation type in the current frame
+ 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]; ///< decoded samples
- uint8_t *bitstream;
- unsigned int bitstream_size;
- unsigned int bitstream_index;
- unsigned int allocated_bitstream_size;
-} FLACContext;
-
-static const int sample_rate_table[] =
-{ 0,
- 88200, 176400, 192000,
- 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
- 0, 0, 0, 0 };
+ int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
-static const int sample_size_table[] =
-{ 0, 8, 12, 0, 16, 20, 24, 0 };
+ FLACDSPContext dsp;
+} FLACContext;
-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);
-int ff_flac_is_extradata_valid(AVCodecContext *avctx,
- enum FLACExtradataFormat *format,
- uint8_t **streaminfo_start)
+static void flac_set_bps(FLACContext *s)
{
- 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;
+ enum AVSampleFormat req = s->avctx->request_sample_fmt;
+ int need32 = s->bps > 16;
+ int want32 = av_get_bytes_per_sample(req) > 2;
+ int planar = av_sample_fmt_is_planar(req);
+
+ if (need32 || want32) {
+ if (planar)
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
+ else
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S32;
+ s->sample_shift = 32 - s->bps;
} 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];
+ if (planar)
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
+ else
+ s->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ s->sample_shift = 16 - s->bps;
}
- return 1;
}
static av_cold int flac_decode_init(AVCodecContext *avctx)
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
- avctx->sample_fmt = 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 (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
+ if (!avpriv_flac_is_extradata_valid(avctx, &format, &streaminfo))
return -1;
/* initialize based on the demuxer-supplied streamdata header */
- ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
+ avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
allocate_buffers(s);
+ flac_set_bps(s);
+ ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps);
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];
+
return 0;
}
assert(s->max_blocksize);
- if (s->max_framesize == 0 && s->max_blocksize) {
- // FIXME header overhead
- s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8;
- }
-
for (i = 0; i < s->channels; i++) {
- s->decoded[i] = av_realloc(s->decoded[i],
- sizeof(int32_t)*s->max_blocksize);
+ s->decoded[i] = av_malloc(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,
- const uint8_t *buffer)
-{
- GetBitContext gb;
- init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
-
- skip_bits(&gb, 16); /* skip min blocksize */
- s->max_blocksize = get_bits(&gb, 16);
- if (s->max_blocksize < 16) {
- 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);
-
- s->samplerate = get_bits_long(&gb, 20);
- s->channels = get_bits(&gb, 3) + 1;
- s->bps = get_bits(&gb, 5) + 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);
-
- skip_bits_long(&gb, 64); /* md5 sum */
- skip_bits_long(&gb, 64); /* md5 sum */
-
- dump_headers(avctx, s);
}
/**
/* need more data */
return 0;
}
- buf += 4;
- metadata_type = bytestream_get_byte(&buf) & 0x7F;
- metadata_size = bytestream_get_be24(&buf);
+ 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;
}
- ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, buf);
+ avpriv_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, &buf[8]);
allocate_buffers(s);
+ flac_set_bps(s);
+ ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
s->got_streaminfo = 1;
return 0;
buf += 4;
do {
- metadata_last = bytestream_get_byte(&buf) & 0x80;
- metadata_size = bytestream_get_be24(&buf);
- if (buf + metadata_size > buf_end) {
+ 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;
}
return buf_size - (buf_end - buf);
}
-static int decode_residuals(FLACContext *s, int channel, int pred_order)
+static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order)
{
int i, tmp, partition, method_type, rice_order;
- int sample = 0, samples;
+ int rice_bits, rice_esc;
+ int samples;
method_type = get_bits(&s->gb, 2);
if (method_type > 1) {
return -1;
}
- sample=
+ rice_bits = 4 + method_type;
+ rice_esc = (1 << rice_bits) - 1;
+
+ decoded += pred_order;
i= pred_order;
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)) {
+ tmp = get_bits(&s->gb, rice_bits);
+ if (tmp == rice_esc) {
tmp = get_bits(&s->gb, 5);
- for (; i < samples; i++, sample++)
- s->decoded[channel][sample] = get_sbits_long(&s->gb, tmp);
+ for (; i < samples; i++)
+ *decoded++ = 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);
+ for (; i < samples; i++) {
+ *decoded++ = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
}
}
i= 0;
return 0;
}
-static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
+static int decode_subframe_fixed(FLACContext *s, int32_t *decoded,
+ int pred_order, int bps)
{
const int blocksize = s->blocksize;
- int32_t *decoded = s->decoded[channel];
- int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i;
+ int a, b, c, d, i;
/* warm up samples */
for (i = 0; i < pred_order; i++) {
- decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
+ decoded[i] = get_sbits_long(&s->gb, bps);
}
- if (decode_residuals(s, channel, pred_order) < 0)
+ if (decode_residuals(s, decoded, pred_order) < 0)
return -1;
if (pred_order > 0)
return 0;
}
-static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
+static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order,
+ int bps)
{
- int i, j;
+ int i;
int coeff_prec, qlevel;
- int coeffs[pred_order];
- int32_t *decoded = s->decoded[channel];
+ int coeffs[32];
/* warm up samples */
for (i = 0; i < pred_order; i++) {
- decoded[i] = get_sbits_long(&s->gb, s->curr_bps);
+ decoded[i] = get_sbits_long(&s->gb, bps);
}
coeff_prec = get_bits(&s->gb, 4) + 1;
}
for (i = 0; i < pred_order; i++) {
- coeffs[i] = get_sbits(&s->gb, coeff_prec);
+ coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec);
}
- if (decode_residuals(s, channel, pred_order) < 0)
+ if (decode_residuals(s, decoded, pred_order) < 0)
return -1;
- if (s->bps > 16) {
- int64_t sum;
- 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) {
- int c;
- int d = decoded[i-pred_order];
- int s0 = 0, s1 = 0;
- for (j = pred_order-1; j > 0; j--) {
- c = coeffs[j];
- s0 += c*d;
- d = decoded[i-j];
- s1 += c*d;
- }
- c = coeffs[0];
- s0 += c*d;
- d = decoded[i] += s0 >> qlevel;
- s1 += c*d;
- decoded[i+1] += s1 >> qlevel;
- }
- if (i < s->blocksize) {
- int sum = 0;
- for (j = 0; j < pred_order; j++)
- sum += coeffs[j] * decoded[i-j-1];
- decoded[i] += sum >> qlevel;
- }
- }
+ s->dsp.lpc(decoded, coeffs, pred_order, qlevel, s->blocksize);
return 0;
}
static inline int decode_subframe(FLACContext *s, int channel)
{
+ int32_t *decoded = s->decoded[channel];
int type, wasted = 0;
+ int bps = s->bps;
int i, tmp;
- s->curr_bps = s->bps;
if (channel == 0) {
- if (s->decorrelation == RIGHT_SIDE)
- s->curr_bps++;
+ if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
+ bps++;
} else {
- if (s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
- s->curr_bps++;
+ if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
+ bps++;
}
if (get_bits1(&s->gb)) {
type = get_bits(&s->gb, 6);
if (get_bits1(&s->gb)) {
+ int left = get_bits_left(&s->gb);
wasted = 1;
+ if ( left < 0 ||
+ (left < bps && !show_bits_long(&s->gb, left)) ||
+ !show_bits_long(&s->gb, bps)) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Invalid number of wasted bits > available bits (%d) - left=%d\n",
+ bps, left);
+ return AVERROR_INVALIDDATA;
+ }
while (!get_bits1(&s->gb))
wasted++;
- s->curr_bps -= wasted;
+ bps -= wasted;
}
- if (s->curr_bps > 32) {
- ff_log_missing_feature(s->avctx, "decorrelated bit depth > 32", 0);
- return -1;
+ if (bps > 32) {
+ av_log_missing_feature(s->avctx, "Decorrelated bit depth > 32", 0);
+ return AVERROR_PATCHWELCOME;
}
//FIXME use av_log2 for types
if (type == 0) {
- tmp = get_sbits_long(&s->gb, s->curr_bps);
+ tmp = get_sbits_long(&s->gb, bps);
for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] = tmp;
+ decoded[i] = tmp;
} else if (type == 1) {
for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] = get_sbits_long(&s->gb, s->curr_bps);
+ decoded[i] = get_sbits_long(&s->gb, bps);
} else if ((type >= 8) && (type <= 12)) {
- if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
+ if (decode_subframe_fixed(s, decoded, type & ~0x8, bps) < 0)
return -1;
} else if (type >= 32) {
- if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
+ if (decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps) < 0)
return -1;
} else {
av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
if (wasted) {
int i;
for (i = 0; i < s->blocksize; i++)
- s->decoded[channel][i] <<= wasted;
+ decoded[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;
}
- 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;
+ 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;
}
- if (blocksize_code == 0) {
- av_log(s->avctx, AV_LOG_ERROR, "reserved blocksize code: 0\n");
- return -1;
- } 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) {
+ s->bps = s->avctx->bits_per_raw_sample = fi.bps;
+ flac_set_bps(s);
}
- if (blocksize * s->channels * (s->is32 ? 4 : 2) > 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);
+ ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps);
+ s->got_streaminfo = 1;
+ dump_headers(s->avctx, (FLACStreaminfo *)s);
+ }
// dump_headers(s->avctx, (FLACStreaminfo *)s);
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 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 bytes_read = 0;
+ 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 (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);
-
- 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;
- }
+ s->max_framesize =
+ ff_flac_get_max_frame_size(s->max_blocksize ? s->max_blocksize : FLAC_MAX_BLOCKSIZE,
+ FLAC_MAX_CHANNELS, 32);
}
/* check that there is at least the smallest decodable amount of data.
- this amount corresponds to the smallest valid FLAC frame possible. */
- if (buf_size < 24)
- goto end;
+ 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;
/* check for inline header */
if (AV_RB32(buf) == MKBETAG('f','L','a','C')) {
av_log(s->avctx, AV_LOG_ERROR, "invalid header\n");
return -1;
}
- bytes_read = get_metadata_size(buf, buf_size);
- goto end;
- }
-
- /* check for frame sync code and resync stream if necessary */
- if ((AV_RB16(buf) & 0xFFFE) != 0xFFF8) {
- const uint8_t *buf_end = buf + buf_size;
- av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
- while (buf+2 < buf_end && (AV_RB16(buf) & 0xFFFE) != 0xFFF8)
- buf++;
- bytes_read = buf_size - (buf_end - buf);
- goto end; // we may not have enough bits left to decode a frame, so try next time
+ return get_metadata_size(buf, buf_size);
}
/* 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;
}
+ bytes_read = (get_bits_count(&s->gb)+7)/8;
-#define DECORRELATE(left, right)\
- assert(s->channels == 2);\
- for (i = 0; i < s->blocksize; i++) {\
- int a= s->decoded[0][i];\
- int b= s->decoded[1][i];\
- 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++) {
- 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)
+ /* 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;
}
- *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
+ s->dsp.decorrelate[s->ch_mode](s->frame.data, s->decoded, s->channels,
+ s->blocksize, s->sample_shift);
- bytes_read = (get_bits_count(&s->gb)+7)/8;
-end:
if (bytes_read > buf_size) {
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
- s->bitstream_size=0;
- s->bitstream_index=0;
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 += bytes_read;
- s->bitstream_size -= bytes_read;
- return input_buf_size;
- } else
- return bytes_read;
+ return bytes_read;
}
static av_cold int flac_decode_close(AVCodecContext *avctx)
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 = AV_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)"),
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32,
+ AV_SAMPLE_FMT_S32P,
+ -1 },
};
projects / ffmpeg / blobdiff