view alac.c @ 10936:b2ea6b0d17bf libavcodec

Update libx264.c to use new libx264 features With b_keyframe instead of IDR for detecting keyframes, ffmpeg should now support periodic encoding with periodic intra refresh (although there is no interface option for it yet). Set the new timebase values for full VFR input support. Bump configure to check for API version 83.

/*
 * ALAC (Apple Lossless Audio Codec) decoder
 * Copyright (c) 2005 David Hammerton
 *
 * 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/alac.c
 * ALAC (Apple Lossless Audio Codec) decoder
 * @author 2005 David Hammerton
 *
 * For more information on the ALAC format, visit:
 *  http://crazney.net/programs/itunes/alac.html
 *
 * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
 * passed through the extradata[_size] fields. This atom is tacked onto
 * the end of an 'alac' stsd atom and has the following format:
 *  bytes 0-3   atom size (0x24), big-endian
 *  bytes 4-7   atom type ('alac', not the 'alac' tag from start of stsd)
 *  bytes 8-35  data bytes needed by decoder
 *
 * Extradata:
 * 32bit  size
 * 32bit  tag (=alac)
 * 32bit  zero?
 * 32bit  max sample per frame
 *  8bit  ?? (zero?)
 *  8bit  sample size
 *  8bit  history mult
 *  8bit  initial history
 *  8bit  kmodifier
 *  8bit  channels?
 * 16bit  ??
 * 32bit  max coded frame size
 * 32bit  bitrate?
 * 32bit  samplerate
 */


#include "avcodec.h"
#include "get_bits.h"
#include "bytestream.h"
#include "unary.h"
#include "mathops.h"

#define ALAC_EXTRADATA_SIZE 36
#define MAX_CHANNELS 2

typedef struct {

    AVCodecContext *avctx;
    GetBitContext gb;
    /* init to 0; first frame decode should initialize from extradata and
     * set this to 1 */
    int context_initialized;

    int numchannels;
    int bytespersample;

    /* buffers */
    int32_t *predicterror_buffer[MAX_CHANNELS];

    int32_t *outputsamples_buffer[MAX_CHANNELS];

    /* stuff from setinfo */
    uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */    /* max samples per frame? */
    uint8_t setinfo_sample_size; /* 0x10 */
    uint8_t setinfo_rice_historymult; /* 0x28 */
    uint8_t setinfo_rice_initialhistory; /* 0x0a */
    uint8_t setinfo_rice_kmodifier; /* 0x0e */
    /* end setinfo stuff */

} ALACContext;

static void allocate_buffers(ALACContext *alac)
{
    int chan;
    for (chan = 0; chan < MAX_CHANNELS; chan++) {
        alac->predicterror_buffer[chan] =
            av_malloc(alac->setinfo_max_samples_per_frame * 4);

        alac->outputsamples_buffer[chan] =
            av_malloc(alac->setinfo_max_samples_per_frame * 4);
    }
}

static int alac_set_info(ALACContext *alac)
{
    const unsigned char *ptr = alac->avctx->extradata;

    ptr += 4; /* size */
    ptr += 4; /* alac */
    ptr += 4; /* 0 ? */

    if(AV_RB32(ptr) >= UINT_MAX/4){
        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
        return -1;
    }

    /* buffer size / 2 ? */
    alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
    ptr++;                          /* ??? */
    alac->setinfo_sample_size           = *ptr++;
    if (alac->setinfo_sample_size > 32) {
        av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n");
        return -1;
    }
    alac->setinfo_rice_historymult      = *ptr++;
    alac->setinfo_rice_initialhistory   = *ptr++;
    alac->setinfo_rice_kmodifier        = *ptr++;
    ptr++;                         /* channels? */
    bytestream_get_be16(&ptr);      /* ??? */
    bytestream_get_be32(&ptr);      /* max coded frame size */
    bytestream_get_be32(&ptr);      /* bitrate ? */
    bytestream_get_be32(&ptr);      /* samplerate */

    allocate_buffers(alac);

    return 0;
}

static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
    /* read x - number of 1s before 0 represent the rice */
    int x = get_unary_0_9(gb);

    if (x > 8) { /* RICE THRESHOLD */
        /* use alternative encoding */
        x = get_bits(gb, readsamplesize);
    } else {
        if (k >= limit)
            k = limit;

        if (k != 1) {
            int extrabits = show_bits(gb, k);

            /* multiply x by 2^k - 1, as part of their strange algorithm */
            x = (x << k) - x;

            if (extrabits > 1) {
                x += extrabits - 1;
                skip_bits(gb, k);
            } else
                skip_bits(gb, k - 1);
        }
    }
    return x;
}

static void bastardized_rice_decompress(ALACContext *alac,
                                 int32_t *output_buffer,
                                 int output_size,
                                 int readsamplesize, /* arg_10 */
                                 int rice_initialhistory, /* arg424->b */
                                 int rice_kmodifier, /* arg424->d */
                                 int rice_historymult, /* arg424->c */
                                 int rice_kmodifier_mask /* arg424->e */
        )
{
    int output_count;
    unsigned int history = rice_initialhistory;
    int sign_modifier = 0;

    for (output_count = 0; output_count < output_size; output_count++) {
        int32_t x;
        int32_t x_modified;
        int32_t final_val;

        /* standard rice encoding */
        int k; /* size of extra bits */

        /* read k, that is bits as is */
        k = av_log2((history >> 9) + 3);
        x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);

        x_modified = sign_modifier + x;
        final_val = (x_modified + 1) / 2;
        if (x_modified & 1) final_val *= -1;

        output_buffer[output_count] = final_val;

        sign_modifier = 0;

        /* now update the history */
        history += x_modified * rice_historymult
                   - ((history * rice_historymult) >> 9);

        if (x_modified > 0xffff)
            history = 0xffff;

        /* special case: there may be compressed blocks of 0 */
        if ((history < 128) && (output_count+1 < output_size)) {
            int k;
            unsigned int block_size;

            sign_modifier = 1;

            k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);

            block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);

            if (block_size > 0) {
                if(block_size >= output_size - output_count){
                    av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count);
                    block_size= output_size - output_count - 1;
                }
                memset(&output_buffer[output_count+1], 0, block_size * 4);
                output_count += block_size;
            }

            if (block_size > 0xffff)
                sign_modifier = 0;

            history = 0;
        }
    }
}

static inline int sign_only(int v)
{
    return v ? FFSIGN(v) : 0;
}

static void predictor_decompress_fir_adapt(int32_t *error_buffer,
                                           int32_t *buffer_out,
                                           int output_size,
                                           int readsamplesize,
                                           int16_t *predictor_coef_table,
                                           int predictor_coef_num,
                                           int predictor_quantitization)
{
    int i;

    /* first sample always copies */
    *buffer_out = *error_buffer;

    if (!predictor_coef_num) {
        if (output_size <= 1)
            return;

        memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
        return;
    }

    if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
      /* second-best case scenario for fir decompression,
       * error describes a small difference from the previous sample only
       */
        if (output_size <= 1)
            return;
        for (i = 0; i < output_size - 1; i++) {
            int32_t prev_value;
            int32_t error_value;

            prev_value = buffer_out[i];
            error_value = error_buffer[i+1];
            buffer_out[i+1] =
                sign_extend((prev_value + error_value), readsamplesize);
        }
        return;
    }

    /* read warm-up samples */
    if (predictor_coef_num > 0)
        for (i = 0; i < predictor_coef_num; i++) {
            int32_t val;

            val = buffer_out[i] + error_buffer[i+1];
            val = sign_extend(val, readsamplesize);
            buffer_out[i+1] = val;
        }

#if 0
    /* 4 and 8 are very common cases (the only ones i've seen). these
     * should be unrolled and optimized
     */
    if (predictor_coef_num == 4) {
        /* FIXME: optimized general case */
        return;
    }

    if (predictor_coef_table == 8) {
        /* FIXME: optimized general case */
        return;
    }
#endif

    /* general case */
    if (predictor_coef_num > 0) {
        for (i = predictor_coef_num + 1; i < output_size; i++) {
            int j;
            int sum = 0;
            int outval;
            int error_val = error_buffer[i];

            for (j = 0; j < predictor_coef_num; j++) {
                sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
                       predictor_coef_table[j];
            }

            outval = (1 << (predictor_quantitization-1)) + sum;
            outval = outval >> predictor_quantitization;
            outval = outval + buffer_out[0] + error_val;
            outval = sign_extend(outval, readsamplesize);

            buffer_out[predictor_coef_num+1] = outval;

            if (error_val > 0) {
                int predictor_num = predictor_coef_num - 1;

                while (predictor_num >= 0 && error_val > 0) {
                    int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
                    int sign = sign_only(val);

                    predictor_coef_table[predictor_num] -= sign;

                    val *= sign; /* absolute value */

                    error_val -= ((val >> predictor_quantitization) *
                                  (predictor_coef_num - predictor_num));

                    predictor_num--;
                }
            } else if (error_val < 0) {
                int predictor_num = predictor_coef_num - 1;

                while (predictor_num >= 0 && error_val < 0) {
                    int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
                    int sign = - sign_only(val);

                    predictor_coef_table[predictor_num] -= sign;

                    val *= sign; /* neg value */

                    error_val -= ((val >> predictor_quantitization) *
                                  (predictor_coef_num - predictor_num));

                    predictor_num--;
                }
            }

            buffer_out++;
        }
    }
}

static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
                                  int16_t *buffer_out,
                                  int numchannels, int numsamples,
                                  uint8_t interlacing_shift,
                                  uint8_t interlacing_leftweight)
{
    int i;
    if (numsamples <= 0)
        return;

    /* weighted interlacing */
    if (interlacing_leftweight) {
        for (i = 0; i < numsamples; i++) {
            int32_t a, b;

            a = buffer[0][i];
            b = buffer[1][i];

            a -= (b * interlacing_leftweight) >> interlacing_shift;
            b += a;

            buffer_out[i*numchannels] = b;
            buffer_out[i*numchannels + 1] = a;
        }

        return;
    }

    /* otherwise basic interlacing took place */
    for (i = 0; i < numsamples; i++) {
        int16_t left, right;

        left = buffer[0][i];
        right = buffer[1][i];

        buffer_out[i*numchannels] = left;
        buffer_out[i*numchannels + 1] = right;
    }
}

static int alac_decode_frame(AVCodecContext *avctx,
                             void *outbuffer, int *outputsize,
                             AVPacket *avpkt)
{
    const uint8_t *inbuffer = avpkt->data;
    int input_buffer_size = avpkt->size;
    ALACContext *alac = avctx->priv_data;

    int channels;
    unsigned int outputsamples;
    int hassize;
    unsigned int readsamplesize;
    int wasted_bytes;
    int isnotcompressed;
    uint8_t interlacing_shift;
    uint8_t interlacing_leftweight;

    /* short-circuit null buffers */
    if (!inbuffer || !input_buffer_size)
        return input_buffer_size;

    /* initialize from the extradata */
    if (!alac->context_initialized) {
        if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
            av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
                ALAC_EXTRADATA_SIZE);
            return input_buffer_size;
        }
        if (alac_set_info(alac)) {
            av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
            return input_buffer_size;
        }
        alac->context_initialized = 1;
    }

    init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);

    channels = get_bits(&alac->gb, 3) + 1;
    if (channels > MAX_CHANNELS) {
        av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
               MAX_CHANNELS);
        return input_buffer_size;
    }

    /* 2^result = something to do with output waiting.
     * perhaps matters if we read > 1 frame in a pass?
     */
    skip_bits(&alac->gb, 4);

    skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */

    /* the output sample size is stored soon */
    hassize = get_bits1(&alac->gb);

    wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */

    /* whether the frame is compressed */
    isnotcompressed = get_bits1(&alac->gb);

    if (hassize) {
        /* now read the number of samples as a 32bit integer */
        outputsamples = get_bits_long(&alac->gb, 32);
        if(outputsamples > alac->setinfo_max_samples_per_frame){
            av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame);
            return -1;
        }
    } else
        outputsamples = alac->setinfo_max_samples_per_frame;

    if(outputsamples > *outputsize / alac->bytespersample){
        av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n");
        return -1;
    }

    *outputsize = outputsamples * alac->bytespersample;
    readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
    if (readsamplesize > MIN_CACHE_BITS) {
        av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize);
        return -1;
    }

    if (!isnotcompressed) {
        /* so it is compressed */
        int16_t predictor_coef_table[channels][32];
        int predictor_coef_num[channels];
        int prediction_type[channels];
        int prediction_quantitization[channels];
        int ricemodifier[channels];
        int i, chan;

        interlacing_shift = get_bits(&alac->gb, 8);
        interlacing_leftweight = get_bits(&alac->gb, 8);

        for (chan = 0; chan < channels; chan++) {
            prediction_type[chan] = get_bits(&alac->gb, 4);
            prediction_quantitization[chan] = get_bits(&alac->gb, 4);

            ricemodifier[chan] = get_bits(&alac->gb, 3);
            predictor_coef_num[chan] = get_bits(&alac->gb, 5);

            /* read the predictor table */
            for (i = 0; i < predictor_coef_num[chan]; i++)
                predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
        }

        if (wasted_bytes)
            av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");

        for (chan = 0; chan < channels; chan++) {
            bastardized_rice_decompress(alac,
                                        alac->predicterror_buffer[chan],
                                        outputsamples,
                                        readsamplesize,
                                        alac->setinfo_rice_initialhistory,
                                        alac->setinfo_rice_kmodifier,
                                        ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
                                        (1 << alac->setinfo_rice_kmodifier) - 1);

            if (prediction_type[chan] == 0) {
                /* adaptive fir */
                predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
                                               alac->outputsamples_buffer[chan],
                                               outputsamples,
                                               readsamplesize,
                                               predictor_coef_table[chan],
                                               predictor_coef_num[chan],
                                               prediction_quantitization[chan]);
            } else {
                av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
                /* I think the only other prediction type (or perhaps this is
                 * just a boolean?) runs adaptive fir twice.. like:
                 * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
                 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
                 * little strange..
                 */
            }
        }
    } else {
        /* not compressed, easy case */
        int i, chan;
        for (i = 0; i < outputsamples; i++)
            for (chan = 0; chan < channels; chan++) {
                int32_t audiobits;

                audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size);

                alac->outputsamples_buffer[chan][i] = audiobits;
            }
        /* wasted_bytes = 0; */
        interlacing_shift = 0;
        interlacing_leftweight = 0;
    }
    if (get_bits(&alac->gb, 3) != 7)
        av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");

    switch(alac->setinfo_sample_size) {
    case 16:
        if (channels == 2) {
            reconstruct_stereo_16(alac->outputsamples_buffer,
                                  (int16_t*)outbuffer,
                                  alac->numchannels,
                                  outputsamples,
                                  interlacing_shift,
                                  interlacing_leftweight);
        } else {
            int i;
            for (i = 0; i < outputsamples; i++) {
                int16_t sample = alac->outputsamples_buffer[0][i];
                ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
            }
        }
        break;
    case 20:
    case 24:
        // It is not clear if there exist any encoder that creates 24 bit ALAC
        // files. iTunes convert 24 bit raw files to 16 bit before encoding.
    case 32:
        av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
        break;
    default:
        break;
    }

    if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
        av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));

    return input_buffer_size;
}

static av_cold int alac_decode_init(AVCodecContext * avctx)
{
    ALACContext *alac = avctx->priv_data;
    alac->avctx = avctx;
    alac->context_initialized = 0;

    alac->numchannels = alac->avctx->channels;
    alac->bytespersample = 2 * alac->numchannels;
    avctx->sample_fmt = SAMPLE_FMT_S16;

    return 0;
}

static av_cold int alac_decode_close(AVCodecContext *avctx)
{
    ALACContext *alac = avctx->priv_data;

    int chan;
    for (chan = 0; chan < MAX_CHANNELS; chan++) {
        av_free(alac->predicterror_buffer[chan]);
        av_free(alac->outputsamples_buffer[chan]);
    }

    return 0;
}

AVCodec alac_decoder = {
    "alac",
    CODEC_TYPE_AUDIO,
    CODEC_ID_ALAC,
    sizeof(ALACContext),
    alac_decode_init,
    NULL,
    alac_decode_close,
    alac_decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
};