0,0 → 1,608 |
/* |
* FLAC (Free Lossless Audio Codec) decoder |
* Copyright (c) 2003 Alex Beregszaszi |
* |
* This file is part of FFmpeg. |
* |
* FFmpeg 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, |
* 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 |
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
*/ |
|
/** |
* @file |
* FLAC (Free Lossless Audio Codec) decoder |
* @author Alex Beregszaszi |
* @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 |
* 34-byte streaminfo structure through avctx->extradata[_size] followed |
* by data starting with the 0xFFF8 marker. |
*/ |
|
#include <limits.h> |
|
#include "libavutil/avassert.h" |
#include "libavutil/channel_layout.h" |
#include "libavutil/crc.h" |
#include "avcodec.h" |
#include "internal.h" |
#include "get_bits.h" |
#include "bytestream.h" |
#include "golomb.h" |
#include "flac.h" |
#include "flacdata.h" |
#include "flacdsp.h" |
#include "thread.h" |
|
typedef struct FLACContext { |
FLACSTREAMINFO |
|
AVCodecContext *avctx; ///< parent AVCodecContext |
GetBitContext gb; ///< GetBitContext initialized to start at the current frame |
|
int blocksize; ///< number of samples in the current frame |
int sample_shift; ///< shift required to make output samples 16-bit or 32-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 |
uint8_t *decoded_buffer; |
unsigned int decoded_buffer_size; |
|
FLACDSPContext dsp; |
} FLACContext; |
|
static int allocate_buffers(FLACContext *s); |
|
static void flac_set_bps(FLACContext *s) |
{ |
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 (planar) |
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P; |
else |
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
s->sample_shift = 16 - s->bps; |
} |
} |
|
static av_cold int flac_decode_init(AVCodecContext *avctx) |
{ |
enum FLACExtradataFormat format; |
uint8_t *streaminfo; |
int ret; |
FLACContext *s = avctx->priv_data; |
s->avctx = avctx; |
|
/* 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 AVERROR_INVALIDDATA; |
|
/* initialize based on the demuxer-supplied streamdata header */ |
avpriv_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo); |
ret = allocate_buffers(s); |
if (ret < 0) |
return ret; |
flac_set_bps(s); |
ff_flacdsp_init(&s->dsp, avctx->sample_fmt, s->bps); |
s->got_streaminfo = 1; |
|
return 0; |
} |
|
static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s) |
{ |
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 int allocate_buffers(FLACContext *s) |
{ |
int buf_size; |
|
av_assert0(s->max_blocksize); |
|
buf_size = av_samples_get_buffer_size(NULL, s->channels, s->max_blocksize, |
AV_SAMPLE_FMT_S32P, 0); |
if (buf_size < 0) |
return buf_size; |
|
av_fast_malloc(&s->decoded_buffer, &s->decoded_buffer_size, buf_size); |
if (!s->decoded_buffer) |
return AVERROR(ENOMEM); |
|
return av_samples_fill_arrays((uint8_t **)s->decoded, NULL, |
s->decoded_buffer, s->channels, |
s->max_blocksize, AV_SAMPLE_FMT_S32P, 0); |
} |
|
/** |
* 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 parse_streaminfo(FLACContext *s, const uint8_t *buf, int buf_size) |
{ |
int metadata_type, metadata_size, ret; |
|
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]); |
ret = allocate_buffers(s); |
if (ret < 0) |
return ret; |
flac_set_bps(s); |
ff_flacdsp_init(&s->dsp, s->avctx->sample_fmt, s->bps); |
s->got_streaminfo = 1; |
|
return 0; |
} |
|
/** |
* 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); |
|
return buf_size - (buf_end - buf); |
} |
|
static int decode_residuals(FLACContext *s, int32_t *decoded, int pred_order) |
{ |
int i, tmp, partition, method_type, rice_order; |
int rice_bits, rice_esc; |
int 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); |
return AVERROR_INVALIDDATA; |
} |
|
rice_order = get_bits(&s->gb, 4); |
|
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); |
return AVERROR_INVALIDDATA; |
} |
|
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, rice_bits); |
if (tmp == rice_esc) { |
tmp = get_bits(&s->gb, 5); |
for (; i < samples; i++) |
*decoded++ = get_sbits_long(&s->gb, tmp); |
} else { |
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, int32_t *decoded, |
int pred_order, int bps) |
{ |
const int blocksize = s->blocksize; |
int av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(d), i; |
int ret; |
|
/* warm up samples */ |
for (i = 0; i < pred_order; i++) { |
decoded[i] = get_sbits_long(&s->gb, bps); |
} |
|
if ((ret = decode_residuals(s, decoded, pred_order)) < 0) |
return ret; |
|
if (pred_order > 0) |
a = decoded[pred_order-1]; |
if (pred_order > 1) |
b = a - decoded[pred_order-2]; |
if (pred_order > 2) |
c = b - decoded[pred_order-2] + decoded[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 AVERROR_INVALIDDATA; |
} |
|
return 0; |
} |
|
static int decode_subframe_lpc(FLACContext *s, int32_t *decoded, int pred_order, |
int bps) |
{ |
int i, ret; |
int coeff_prec, qlevel; |
int coeffs[32]; |
|
/* warm up samples */ |
for (i = 0; i < pred_order; i++) { |
decoded[i] = get_sbits_long(&s->gb, bps); |
} |
|
coeff_prec = get_bits(&s->gb, 4) + 1; |
if (coeff_prec == 16) { |
av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n"); |
return AVERROR_INVALIDDATA; |
} |
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); |
return AVERROR_INVALIDDATA; |
} |
|
for (i = 0; i < pred_order; i++) { |
coeffs[pred_order - i - 1] = get_sbits(&s->gb, coeff_prec); |
} |
|
if ((ret = decode_residuals(s, decoded, pred_order)) < 0) |
return ret; |
|
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, ret; |
|
if (channel == 0) { |
if (s->ch_mode == FLAC_CHMODE_RIGHT_SIDE) |
bps++; |
} else { |
if (s->ch_mode == FLAC_CHMODE_LEFT_SIDE || s->ch_mode == FLAC_CHMODE_MID_SIDE) |
bps++; |
} |
|
if (get_bits1(&s->gb)) { |
av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n"); |
return AVERROR_INVALIDDATA; |
} |
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++; |
bps -= wasted; |
} |
if (bps > 32) { |
avpriv_report_missing_feature(s->avctx, "Decorrelated bit depth > 32"); |
return AVERROR_PATCHWELCOME; |
} |
|
//FIXME use av_log2 for types |
if (type == 0) { |
tmp = get_sbits_long(&s->gb, bps); |
for (i = 0; i < s->blocksize; i++) |
decoded[i] = tmp; |
} else if (type == 1) { |
for (i = 0; i < s->blocksize; i++) |
decoded[i] = get_sbits_long(&s->gb, bps); |
} else if ((type >= 8) && (type <= 12)) { |
if ((ret = decode_subframe_fixed(s, decoded, type & ~0x8, bps)) < 0) |
return ret; |
} else if (type >= 32) { |
if ((ret = decode_subframe_lpc(s, decoded, (type & ~0x20)+1, bps)) < 0) |
return ret; |
} else { |
av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n"); |
return AVERROR_INVALIDDATA; |
} |
|
if (wasted) { |
int i; |
for (i = 0; i < s->blocksize; i++) |
decoded[i] <<= wasted; |
} |
|
return 0; |
} |
|
static int decode_frame(FLACContext *s) |
{ |
int i, ret; |
GetBitContext *gb = &s->gb; |
FLACFrameInfo fi; |
|
if ((ret = ff_flac_decode_frame_header(s->avctx, gb, &fi, 0)) < 0) { |
av_log(s->avctx, AV_LOG_ERROR, "invalid frame header\n"); |
return ret; |
} |
|
if (s->channels && fi.channels != s->channels && s->got_streaminfo) { |
s->channels = s->avctx->channels = fi.channels; |
ff_flac_set_channel_layout(s->avctx); |
ret = allocate_buffers(s); |
if (ret < 0) |
return ret; |
} |
s->channels = s->avctx->channels = fi.channels; |
if (!s->avctx->channel_layout) |
ff_flac_set_channel_layout(s->avctx); |
s->ch_mode = fi.ch_mode; |
|
if (!s->bps && !fi.bps) { |
av_log(s->avctx, AV_LOG_ERROR, "bps not found in STREAMINFO or frame header\n"); |
return AVERROR_INVALIDDATA; |
} |
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 AVERROR_INVALIDDATA; |
} |
|
if (!s->bps) { |
s->bps = s->avctx->bits_per_raw_sample = fi.bps; |
flac_set_bps(s); |
} |
|
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 AVERROR_INVALIDDATA; |
} |
s->blocksize = fi.blocksize; |
|
if (!s->samplerate && !fi.samplerate) { |
av_log(s->avctx, AV_LOG_ERROR, "sample rate not found in STREAMINFO" |
" or frame header\n"); |
return AVERROR_INVALIDDATA; |
} |
if (fi.samplerate == 0) |
fi.samplerate = s->samplerate; |
s->samplerate = s->avctx->sample_rate = fi.samplerate; |
|
if (!s->got_streaminfo) { |
ret = allocate_buffers(s); |
if (ret < 0) |
return ret; |
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); |
|
/* subframes */ |
for (i = 0; i < s->channels; i++) { |
if ((ret = decode_subframe(s, i)) < 0) |
return ret; |
} |
|
align_get_bits(gb); |
|
/* frame footer */ |
skip_bits(gb, 16); /* data crc */ |
|
return 0; |
} |
|
static int flac_decode_frame(AVCodecContext *avctx, void *data, |
int *got_frame_ptr, AVPacket *avpkt) |
{ |
AVFrame *frame = data; |
ThreadFrame tframe = { .f = data }; |
const uint8_t *buf = avpkt->data; |
int buf_size = avpkt->size; |
FLACContext *s = avctx->priv_data; |
int bytes_read = 0; |
int ret; |
|
*got_frame_ptr = 0; |
|
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 (buf_size > 5 && !memcmp(buf, "\177FLAC", 5)) { |
av_log(s->avctx, AV_LOG_DEBUG, "skiping flac header packet 1\n"); |
return buf_size; |
} |
|
if (buf_size > 0 && (*buf & 0x7F) == FLAC_METADATA_TYPE_VORBIS_COMMENT) { |
av_log(s->avctx, AV_LOG_DEBUG, "skiping vorbis comment\n"); |
return 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; |
|
/* check for inline header */ |
if (AV_RB32(buf) == MKBETAG('f','L','a','C')) { |
if (!s->got_streaminfo && (ret = parse_streaminfo(s, buf, buf_size))) { |
av_log(s->avctx, AV_LOG_ERROR, "invalid header\n"); |
return ret; |
} |
return get_metadata_size(buf, buf_size); |
} |
|
/* decode frame */ |
if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0) |
return ret; |
if ((ret = decode_frame(s)) < 0) { |
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n"); |
return ret; |
} |
bytes_read = get_bits_count(&s->gb)/8; |
|
if ((s->avctx->err_recognition & (AV_EF_CRCCHECK|AV_EF_COMPLIANT)) && |
av_crc(av_crc_get_table(AV_CRC_16_ANSI), |
0, buf, bytes_read)) { |
av_log(s->avctx, AV_LOG_ERROR, "CRC error at PTS %"PRId64"\n", avpkt->pts); |
if (s->avctx->err_recognition & AV_EF_EXPLODE) |
return AVERROR_INVALIDDATA; |
} |
|
/* get output buffer */ |
frame->nb_samples = s->blocksize; |
if ((ret = ff_thread_get_buffer(avctx, &tframe, 0)) < 0) |
return ret; |
|
s->dsp.decorrelate[s->ch_mode](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 AVERROR_INVALIDDATA; |
} |
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; |
|
return bytes_read; |
} |
|
static int init_thread_copy(AVCodecContext *avctx) |
{ |
FLACContext *s = avctx->priv_data; |
s->decoded_buffer = NULL; |
s->decoded_buffer_size = 0; |
s->avctx = avctx; |
if (s->max_blocksize) |
return allocate_buffers(s); |
return 0; |
} |
|
static av_cold int flac_decode_close(AVCodecContext *avctx) |
{ |
FLACContext *s = avctx->priv_data; |
|
av_freep(&s->decoded_buffer); |
|
return 0; |
} |
|
AVCodec ff_flac_decoder = { |
.name = "flac", |
.long_name = NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"), |
.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, |
.init_thread_copy = ONLY_IF_THREADS_ENABLED(init_thread_copy), |
.capabilities = CODEC_CAP_DR1 | CODEC_CAP_FRAME_THREADS, |
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, |
AV_SAMPLE_FMT_S16P, |
AV_SAMPLE_FMT_S32, |
AV_SAMPLE_FMT_S32P, |
AV_SAMPLE_FMT_NONE }, |
}; |