* 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
* 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 "golomb.h"
#include "flac.h"
#include "flacdata.h"
+#include "flacdsp.h"
#undef NDEBUG
#include <assert.h>
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
int ch_mode; ///< channel decorrelation type in the current frame
int got_streaminfo; ///< indicates if the STREAMINFO has been read
int32_t *decoded[FLAC_MAX_CHANNELS]; ///< decoded samples
+
+ FLACDSPContext dsp;
} FLACContext;
+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 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 = 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 (!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);
- if (s->bps > 16)
- avctx->sample_fmt = AV_SAMPLE_FMT_S32;
- else
- avctx->sample_fmt = AV_SAMPLE_FMT_S16;
+ 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);
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);
}
}
-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 < 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);
-
- 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;
-
- 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);
-}
-
-void ff_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 the STREAMINFO from an inline header.
* @param s the flac decoding context
/* need more data */
return 0;
}
- ff_flac_parse_block_header(&buf[4], NULL, &metadata_type, &metadata_size);
+ 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[8]);
+ 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 {
- ff_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
+ if (buf_end - buf < 4)
+ return 0;
+ avpriv_flac_parse_block_header(buf, &metadata_last, NULL, &metadata_size);
buf += 4;
- if (buf + metadata_size > buf_end) {
+ 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[32];
- int32_t *decoded = s->decoded[channel];
/* 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->ch_mode == FLAC_CHMODE_RIGHT_SIDE)
- s->curr_bps++;
+ bps++;
} else {
if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE)
- s->curr_bps++;
+ 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) {
- av_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;
"supported\n");
return -1;
}
- s->bps = s->avctx->bits_per_raw_sample = fi.bps;
- 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 (!s->bps) {
+ s->bps = s->avctx->bits_per_raw_sample = fi.bps;
+ flac_set_bps(s);
}
if (!s->max_blocksize)
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);
}
return 0;
}
-static int flac_decode_frame(AVCodecContext *avctx,
- void *data, int *data_size,
- AVPacket *avpkt)
+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, bytes_read = 0;
- int16_t *samples_16 = data;
- int32_t *samples_32 = data;
- int alloc_data_size= *data_size;
- int output_size;
+ int bytes_read = 0;
+ int ret;
- *data_size=0;
+ *got_frame_ptr = 0;
if (s->max_framesize == 0) {
s->max_framesize =
/* 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 < 11)
+ if (buf_size < FLAC_MIN_FRAME_SIZE)
return buf_size;
/* check for inline header */
}
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");
- return -1;
- }
- *data_size = output_size;
-
-#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->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 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)
+ /* 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;
}
+ s->dsp.decorrelate[s->ch_mode](s->frame.data, s->decoded, s->channels,
+ s->blocksize, s->sample_shift);
+
if (bytes_read > buf_size) {
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", bytes_read - buf_size);
return -1;
buf_size - bytes_read, buf_size);
}
+ *got_frame_ptr = 1;
+ *(AVFrame *)data = s->frame;
+
return bytes_read;
}
return 0;
}
-AVCodec flac_decoder = {
- "flac",
- AVMEDIA_TYPE_AUDIO,
- CODEC_ID_FLAC,
- sizeof(FLACContext),
- flac_decode_init,
- NULL,
- flac_decode_close,
- flac_decode_frame,
- .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