/* * Bink Audio decoder * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org) * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu) * * 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 * Bink Audio decoder * * Technical details here: *  http://wiki.multimedia.cx/index.php?title=Bink_Audio */ #include "config_components.h" #include "libavutil/channel_layout.h"#include "libavutil/intfloat.h"#include "libavutil/mem_internal.h"#include "libavutil/tx.h" #define BITSTREAM_READER_LE#include "avcodec.h"#include "decode.h"#include "get_bits.h"#include "codec_internal.h"#include "internal.h"#include "wma_freqs.h" #define MAX_DCT_CHANNELS 6#define MAX_CHANNELS 2#define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11) typedef struct BinkAudioContext {    GetBitContext gb;    int version_b;          ///< Bink version 'b'    int first;    int channels;    int ch_offset;    int frame_len;          ///< transform size (samples)    int overlap_len;        ///< overlap size (samples)    int block_size;    int num_bands;    float root;    unsigned int bands[26];    float previous[MAX_DCT_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block    float quant_table[96];    AVPacket *pkt;    AVTXContext *tx;    av_tx_fn tx_fn;} BinkAudioContext;  static av_cold int decode_init(AVCodecContext *avctx){    BinkAudioContext *s = avctx->priv_data;    int sample_rate = avctx->sample_rate;    int sample_rate_half;    int i, ret;    int frame_len_bits;    int max_channels = avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT ? MAX_CHANNELS : MAX_DCT_CHANNELS;    int channels = avctx->ch_layout.nb_channels;     /* determine frame length */    if (avctx->sample_rate < 22050) {        frame_len_bits = 9;    } else if (avctx->sample_rate < 44100) {        frame_len_bits = 10;    } else {        frame_len_bits = 11;    }     if (channels < 1 || channels > max_channels) {        av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", channels);        return AVERROR_INVALIDDATA;    }    av_channel_layout_uninit(&avctx->ch_layout);    av_channel_layout_default(&avctx->ch_layout, channels);     s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';     if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {        // audio is already interleaved for the RDFT format variant        avctx->sample_fmt = AV_SAMPLE_FMT_FLT;        if (sample_rate > INT_MAX / channels)            return AVERROR_INVALIDDATA;        sample_rate  *= channels;        s->channels = 1;        if (!s->version_b)            frame_len_bits += av_log2(channels);    } else {        s->channels = channels;        avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;    }     s->frame_len     = 1 << frame_len_bits;    s->overlap_len   = s->frame_len / 16;    s->block_size    = (s->frame_len - s->overlap_len) * FFMIN(MAX_CHANNELS, s->channels);    sample_rate_half = (sample_rate + 1LL) / 2;    if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)        s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);    else        s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);    for (i = 0; i < 96; i++) {        /* constant is result of 0.066399999/log10(M_E) */        s->quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;    }     /* calculate number of bands */    for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)        if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])            break;     /* populate bands data */    s->bands[0] = 2;    for (i = 1; i < s->num_bands; i++)        s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;    s->bands[s->num_bands] = s->frame_len;     s->first = 1;     if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {        float scale = 0.5;        ret = av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_RDFT, 1, 1 << frame_len_bits, &scale, 0);    } else if (CONFIG_BINKAUDIO_DCT_DECODER) {        float scale = 1.0 / (1 << frame_len_bits);        ret = av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_DCT, 1, 1 << (frame_len_bits - 1), &scale, 0);    } else {        av_assert0(0);    }    if (ret < 0)        return ret;     s->pkt = avctx->internal->in_pkt;     return 0;} static float get_float(GetBitContext *gb){    int power = get_bits(gb, 5);    float f = ldexpf(get_bits(gb, 23), power - 23);    if (get_bits1(gb))        f = -f;    return f;} static const uint8_t rle_length_tab[16] = {    2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64}; /** * Decode Bink Audio block * @param[out] out Output buffer (must contain s->block_size elements) * @return 0 on success, negative error code on failure */static int decode_block(BinkAudioContext *s, float **out, int use_dct,                        int channels, int ch_offset){    int ch, i, j, k;    float q, quant[25];    int width, coeff;    GetBitContext *gb = &s->gb;    LOCAL_ALIGNED_32(float, coeffs, [4098]);     if (use_dct)        skip_bits(gb, 2);     for (ch = 0; ch < channels; ch++) {        if (s->version_b) {            if (get_bits_left(gb) < 64)                return AVERROR_INVALIDDATA;            coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;            coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;        } else {            if (get_bits_left(gb) < 58)                return AVERROR_INVALIDDATA;            coeffs[0] = get_float(gb) * s->root;            coeffs[1] = get_float(gb) * s->root;        }         if (get_bits_left(gb) < s->num_bands * 8)            return AVERROR_INVALIDDATA;        for (i = 0; i < s->num_bands; i++) {            int value = get_bits(gb, 8);            quant[i]  = s->quant_table[FFMIN(value, 95)];        }         k = 0;        q = quant[0];         // parse coefficients        i = 2;        while (i < s->frame_len) {            if (s->version_b) {                j = i + 16;            } else {                int v = get_bits1(gb);                if (v) {                    v = get_bits(gb, 4);                    j = i + rle_length_tab[v] * 8;                } else {                    j = i + 8;                }            }             j = FFMIN(j, s->frame_len);             width = get_bits(gb, 4);            if (width == 0) {                memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));                i = j;                while (s->bands[k] < i)                    q = quant[k++];            } else {                while (i < j) {                    if (s->bands[k] == i)                        q = quant[k++];                    coeff = get_bits(gb, width);                    if (coeff) {                        int v;                        v = get_bits1(gb);                        if (v)                            coeffs[i] = -q * coeff;                        else                            coeffs[i] =  q * coeff;                    } else {                        coeffs[i] = 0.0f;                    }                    i++;                }            }        }         if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {            coeffs[0] /= 0.5;            s->tx_fn(s->tx, out[ch + ch_offset], coeffs, sizeof(float));        } else if (CONFIG_BINKAUDIO_RDFT_DECODER) {            for (int i = 2; i < s->frame_len; i += 2)                coeffs[i + 1] *= -1;             coeffs[s->frame_len + 0] = coeffs[1];            coeffs[s->frame_len + 1] = coeffs[1] = 0;            s->tx_fn(s->tx, out[ch + ch_offset], coeffs, sizeof(AVComplexFloat));        }    }     for (ch = 0; ch < channels; ch++) {        int j;        int count = s->overlap_len * channels;        if (!s->first) {            j = ch;            for (i = 0; i < s->overlap_len; i++, j += channels)                out[ch + ch_offset][i] = (s->previous[ch + ch_offset][i] * (count - j) +                                                  out[ch + ch_offset][i] *          j) / count;        }        memcpy(s->previous[ch + ch_offset], &out[ch + ch_offset][s->frame_len - s->overlap_len],               s->overlap_len * sizeof(*s->previous[ch + ch_offset]));    }     s->first = 0;     return 0;} static av_cold int decode_end(AVCodecContext *avctx){    BinkAudioContext * s = avctx->priv_data;    av_tx_uninit(&s->tx);    return 0;} static void get_bits_align32(GetBitContext *s){    int n = (-get_bits_count(s)) & 31;    if (n) skip_bits(s, n);} static int binkaudio_receive_frame(AVCodecContext *avctx, AVFrame *frame){    BinkAudioContext *s = avctx->priv_data;    GetBitContext *gb = &s->gb;    int new_pkt, ret; again:    new_pkt = !s->pkt->data;    if (!s->pkt->data) {        ret = ff_decode_get_packet(avctx, s->pkt);        if (ret < 0) {            s->ch_offset = 0;            return ret;        }         if (s->pkt->size < 4) {            av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");            ret = AVERROR_INVALIDDATA;            goto fail;        }         ret = init_get_bits8(gb, s->pkt->data, s->pkt->size);        if (ret < 0)            goto fail;         /* skip reported size */        skip_bits_long(gb, 32);    }     /* get output buffer */    if (s->ch_offset == 0) {        frame->nb_samples = s->frame_len;        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)            goto fail;        if (!new_pkt)            frame->pts = AV_NOPTS_VALUE;    }     if (decode_block(s, (float **)frame->extended_data,                     avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT,                     FFMIN(MAX_CHANNELS, s->channels - s->ch_offset), s->ch_offset)) {        av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");        ret = AVERROR_INVALIDDATA;        goto fail;    }    s->ch_offset += MAX_CHANNELS;    get_bits_align32(gb);    if (!get_bits_left(gb)) {        memset(gb, 0, sizeof(*gb));        av_packet_unref(s->pkt);    }    if (s->ch_offset >= s->channels) {        s->ch_offset = 0;    } else {        goto again;    }     frame->nb_samples = s->block_size / FFMIN(avctx->ch_layout.nb_channels, MAX_CHANNELS);     return 0;fail:    s->ch_offset = 0;    av_packet_unref(s->pkt);    return ret;} static void decode_flush(AVCodecContext *avctx){    BinkAudioContext *const s = avctx->priv_data;     /* s->pkt coincides with avctx->internal->in_pkt     * and is unreferenced generically when flushing. */    s->first = 1;    s->ch_offset = 0;} const FFCodec ff_binkaudio_rdft_decoder = {    .p.name         = "binkaudio_rdft",    CODEC_LONG_NAME("Bink Audio (RDFT)"),    .p.type         = AVMEDIA_TYPE_AUDIO,    .p.id           = AV_CODEC_ID_BINKAUDIO_RDFT,    .priv_data_size = sizeof(BinkAudioContext),    .init           = decode_init,    .flush          = decode_flush,    .close          = decode_end,    FF_CODEC_RECEIVE_FRAME_CB(binkaudio_receive_frame),    .p.capabilities = AV_CODEC_CAP_DR1,    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,}; const FFCodec ff_binkaudio_dct_decoder = {    .p.name         = "binkaudio_dct",    CODEC_LONG_NAME("Bink Audio (DCT)"),    .p.type         = AVMEDIA_TYPE_AUDIO,    .p.id           = AV_CODEC_ID_BINKAUDIO_DCT,    .priv_data_size = sizeof(BinkAudioContext),    .init           = decode_init,    .flush          = decode_flush,    .close          = decode_end,    FF_CODEC_RECEIVE_FRAME_CB(binkaudio_receive_frame),    .p.capabilities = AV_CODEC_CAP_DR1,    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,};