annotate flacdec.c @ 8991:ca768cb2bfb6 libavcodec

Use last decoded SPS as current SPS in order to parse picture timing SEI correctly. This works around an apparent H.264 standard deficiency. Patch by Ivan Schreter, schreter gmx net

/*
 * 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 libavcodec/flacdec.c
 * FLAC (Free Lossless Audio Codec) decoder
 * @author Alex Beregszaszi
 *
 * For more information on the FLAC format, visit:
 *  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>

#define ALT_BITSTREAM_READER
#include "libavutil/crc.h"
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"
#include "flac.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
    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

    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 };

static const int sample_size_table[] =
{ 0, 8, 12, 0, 16, 20, 24, 0 };

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 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);
static int metadata_parse(FLACContext *s);

static av_cold int flac_decode_init(AVCodecContext *avctx)
{
    FLACContext *s = avctx->priv_data;
    s->avctx = avctx;

    avctx->sample_fmt = SAMPLE_FMT_S16;

    if (avctx->extradata_size > 4) {
        /* initialize based on the demuxer-supplied streamdata header */
        if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
            ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s,
                                     avctx->extradata);
            allocate_buffers(s);
        } else {
            init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
            metadata_parse(s);
        }
    }

    return 0;
}

static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
{
    av_log(avctx, AV_LOG_DEBUG, "  Blocksize: %d .. %d\n", s->min_blocksize,
           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 void allocate_buffers(FLACContext *s)
{
    int i;

    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);
    }

    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);

    /* mandatory streaminfo */
    s->min_blocksize = get_bits(&gb, 16);
    s->max_blocksize = get_bits(&gb, 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_long(&gb, 4);

    skip_bits(&gb, 64); /* md5 sum */
    skip_bits(&gb, 64); /* md5 sum */

    dump_headers(avctx, s);
}

/**
 * Parse a list of metadata blocks. This list of blocks must begin with
 * the fLaC marker.
 * @param s the flac decoding context containing the gb bit reader used to
 *          parse metadata
 * @return 1 if some metadata was read, 0 if no fLaC marker was found
 */
static int metadata_parse(FLACContext *s)
{
    int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0;
    int initial_pos= get_bits_count(&s->gb);

    if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
        skip_bits(&s->gb, 32);

        do {
            metadata_last = get_bits1(&s->gb);
            metadata_type = get_bits(&s->gb, 7);
            metadata_size = get_bits_long(&s->gb, 24);

            if (get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits) {
                skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb));
                break;
            }

            if (metadata_size) {
                switch (metadata_type) {
                case FLAC_METADATA_TYPE_STREAMINFO:
                    ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s,
                                             s->gb.buffer+get_bits_count(&s->gb)/8);
                    streaminfo_updated = 1;

                default:
                    for (i = 0; i < metadata_size; i++)
                        skip_bits(&s->gb, 8);
                }
            }
        } while (!metadata_last);

        if (streaminfo_updated)
            allocate_buffers(s);
        return 1;
    }
    return 0;
}

static int decode_residuals(FLACContext *s, int channel, int pred_order)
{
    int i, tmp, partition, method_type, rice_order;
    int sample = 0, 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 -1;
    }

    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 -1;
    }

    sample=
    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, 5);
            for (; i < samples; i++, sample++)
                s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
        } else {
            for (; i < samples; i++, sample++) {
                s->decoded[channel][sample] = 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)
{
    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;

    /* warm up samples */
    for (i = 0; i < pred_order; i++) {
        decoded[i] = get_sbits(&s->gb, s->curr_bps);
    }

    if (decode_residuals(s, channel, pred_order) < 0)
        return -1;

    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 -1;
    }

    return 0;
}

static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
{
    int i, j;
    int coeff_prec, qlevel;
    int coeffs[pred_order];
    int32_t *decoded = s->decoded[channel];

    /* warm up samples */
    for (i = 0; i < pred_order; i++) {
        decoded[i] = get_sbits(&s->gb, s->curr_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 -1;
    }
    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 -1;
    }

    for (i = 0; i < pred_order; i++) {
        coeffs[i] = get_sbits(&s->gb, coeff_prec);
    }

    if (decode_residuals(s, channel, 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;
        }
    }

    return 0;
}

static inline int decode_subframe(FLACContext *s, int channel)
{
    int type, wasted = 0;
    int i, tmp;

    s->curr_bps = s->bps;
    if (channel == 0) {
        if (s->decorrelation == RIGHT_SIDE)
            s->curr_bps++;
    } else {
        if (s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
            s->curr_bps++;
    }

    if (get_bits1(&s->gb)) {
        av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
        return -1;
    }
    type = get_bits(&s->gb, 6);

    if (get_bits1(&s->gb)) {
        wasted = 1;
        while (!get_bits1(&s->gb))
            wasted++;
        s->curr_bps -= wasted;
    }

//FIXME use av_log2 for types
    if (type == 0) {
        tmp = get_sbits(&s->gb, s->curr_bps);
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] = tmp;
    } else if (type == 1) {
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
    } else if ((type >= 8) && (type <= 12)) {
        if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
            return -1;
    } else if (type >= 32) {
        if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
            return -1;
    } else {
        av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
        return -1;
    }

    if (wasted) {
        int i;
        for (i = 0; i < s->blocksize; i++)
            s->decoded[channel][i] <<= wasted;
    }

    return 0;
}

static int decode_frame(FLACContext *s, int alloc_data_size)
{
    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);

    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);
        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);
        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;

    if (get_bits1(&s->gb)) {
        av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
        return -1;
    }

    if (get_utf8(&s->gb) < 0) {
        av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
        return -1;
    }

    if (blocksize_code == 0)
        blocksize = s->min_blocksize;
    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 (blocksize * s->channels * sizeof(int16_t) > 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);
        return -1;
    }

    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);
        return -1;
    }

    s->blocksize    = blocksize;
    s->samplerate   = samplerate;
    s->bps          = bps;
    s->decorrelation= decorrelation;

//    dump_headers(s->avctx, (FLACStreaminfo *)s);

    /* subframes */
    for (i = 0; i < s->channels; i++) {
        if (decode_subframe(s, i) < 0)
            return -1;
    }

    align_get_bits(&s->gb);

    /* frame footer */
    skip_bits(&s->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)
{
    FLACContext *s = avctx->priv_data;
    int tmp = 0, i, j = 0, input_buf_size = 0;
    int16_t *samples_16 = data;
    int32_t *samples_32 = data;
    int alloc_data_size= *data_size;

    *data_size=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;
        }
    }

    init_get_bits(&s->gb, buf, buf_size*8);

    if (metadata_parse(s))
        goto end;

    tmp = show_bits(&s->gb, 16);
    if ((tmp & 0xFFFE) != 0xFFF8) {
        av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
        while (get_bits_count(&s->gb)/8+2 < buf_size && (show_bits(&s->gb, 16) & 0xFFFE) != 0xFFF8)
            skip_bits(&s->gb, 8);
        goto end; // we may not have enough bits left to decode a frame, so try next time
    }
    skip_bits(&s->gb, 16);
    if (decode_frame(s, alloc_data_size) < 0) {
        av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
        s->bitstream_size=0;
        s->bitstream_index=0;
        return -1;
    }

#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)
    }

    *data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);

end:
    i= (get_bits_count(&s->gb)+7)/8;
    if (i > buf_size) {
        av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
        s->bitstream_size=0;
        s->bitstream_index=0;
        return -1;
    }

    if (s->bitstream_size) {
        s->bitstream_index += i;
        s->bitstream_size  -= i;
        return input_buf_size;
    } else
        return i;
}

static av_cold int flac_decode_close(AVCodecContext *avctx)
{
    FLACContext *s = avctx->priv_data;
    int i;

    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)"),
};